res_pjsip_session.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Mark Michelson <mmichelson@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
 
/*** MODULEINFO
    <depend>pjproject</depend>
    <depend>res_pjsip</depend>
    <support_level>core</support_level>
 ***/
 
#include "asterisk.h"
 
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjlib.h>
#include <pjmedia.h>
 
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/res_pjsip_session_caps.h"
#include "asterisk/callerid.h"
#include "asterisk/datastore.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/astobj2.h"
#include "asterisk/lock.h"
#include "asterisk/uuid.h"
#include "asterisk/pbx.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/causes.h"
#include "asterisk/sdp_srtp.h"
#include "asterisk/dsp.h"
#include "asterisk/acl.h"
#include "asterisk/features_config.h"
#include "asterisk/pickup.h"
#include "asterisk/test.h"
#include "asterisk/stream.h"
#include "asterisk/vector.h"
 
#include "res_pjsip_session/pjsip_session.h"
 
#define SDP_HANDLER_BUCKETS 11
 
#define MOD_DATA_ON_RESPONSE "on_response"
#define MOD_DATA_NAT_HOOK "nat_hook"
 
/* Most common case is one audio and one video stream */
#define DEFAULT_NUM_SESSION_MEDIA 2
 
/* Some forward declarations */
static void handle_session_begin(struct ast_sip_session *session);
static void handle_session_end(struct ast_sip_session *session);
static void handle_session_destroy(struct ast_sip_session *session);
static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata);
static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata,
        enum ast_sip_session_response_priority response_priority);
static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
        enum ast_sip_session_response_priority response_priority);
static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata);
static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata);
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 *pending_media_state,
        struct ast_sip_session_media_state *active_media_state,
        int queued);
 
/*! \brief NAT hook for modifying outgoing messages with SDP */
static struct ast_sip_nat_hook *nat_hook;
 
/*!
 * \brief Registered SDP stream handlers
 *
 * This container is keyed on stream types. Each
 * object in the container is a linked list of
 * handlers for the stream type.
 */
static struct ao2_container *sdp_handlers;
 
/*!
 * These are the objects in the sdp_handlers container
 */
struct sdp_handler_list {
    /* The list of handlers to visit */
    AST_LIST_HEAD_NOLOCK(, ast_sip_session_sdp_handler) list;
    /* The handlers in this list handle streams of this type */
    char stream_type[1];
};
 
static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer, const unsigned int ignore_active_stream_topology);
 
static int sdp_handler_list_hash(const void *obj, int flags)
{
    const struct sdp_handler_list *handler_list = obj;
    const char *stream_type = flags & OBJ_KEY ? obj : handler_list->stream_type;
 
    return ast_str_hash(stream_type);
}
 
const char *ast_sip_session_get_name(const struct ast_sip_session *session)
{
    if (!session) {
        return "(null session)";
    }
    if (session->channel) {
        return ast_channel_name(session->channel);
    } else if (session->endpoint) {
        return ast_sorcery_object_get_id(session->endpoint);
    } else {
        return "unknown";
    }
}
 
int ast_sip_can_present_connected_id(const struct ast_sip_session *session, const struct ast_party_id *id)
{
    return id->number.valid
        && (session->endpoint->id.trust_outbound
           || (ast_party_id_presentation(id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED);
}
 
static int sdp_handler_list_cmp(void *obj, void *arg, int flags)
{
    struct sdp_handler_list *handler_list1 = obj;
    struct sdp_handler_list *handler_list2 = arg;
    const char *stream_type2 = flags & OBJ_KEY ? arg : handler_list2->stream_type;
 
    return strcmp(handler_list1->stream_type, stream_type2) ? 0 : CMP_MATCH | CMP_STOP;
}
 
int ast_sip_session_register_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type)
{
    RAII_VAR(struct sdp_handler_list *, handler_list,
            ao2_find(sdp_handlers, stream_type, OBJ_KEY), ao2_cleanup);
    SCOPED_AO2LOCK(lock, sdp_handlers);
 
    if (handler_list) {
        struct ast_sip_session_sdp_handler *iter;
        /* Check if this handler is already registered for this stream type */
        AST_LIST_TRAVERSE(&handler_list->list, iter, next) {
            if (!strcmp(iter->id, handler->id)) {
                ast_log(LOG_WARNING, "Handler '%s' already registered for stream type '%s'.\n", handler->id, stream_type);
                return -1;
            }
        }
        AST_LIST_INSERT_TAIL(&handler_list->list, handler, next);
        ast_debug(1, "Registered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
 
        return 0;
    }
 
    /* No stream of this type has been registered yet, so we need to create a new list */
    handler_list = ao2_alloc(sizeof(*handler_list) + strlen(stream_type), NULL);
    if (!handler_list) {
        return -1;
    }
    /* Safe use of strcpy */
    strcpy(handler_list->stream_type, stream_type);
    AST_LIST_HEAD_INIT_NOLOCK(&handler_list->list);
    AST_LIST_INSERT_TAIL(&handler_list->list, handler, next);
    if (!ao2_link(sdp_handlers, handler_list)) {
        return -1;
    }
    ast_debug(1, "Registered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
 
    return 0;
}
 
static int remove_handler(void *obj, void *arg, void *data, int flags)
{
    struct sdp_handler_list *handler_list = obj;
    struct ast_sip_session_sdp_handler *handler = data;
    struct ast_sip_session_sdp_handler *iter;
    const char *stream_type = arg;
 
    AST_LIST_TRAVERSE_SAFE_BEGIN(&handler_list->list, iter, next) {
        if (!strcmp(iter->id, handler->id)) {
            AST_LIST_REMOVE_CURRENT(next);
            ast_debug(1, "Unregistered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
        }
    }
    AST_LIST_TRAVERSE_SAFE_END;
 
    if (AST_LIST_EMPTY(&handler_list->list)) {
        ast_debug(3, "No more handlers exist for stream type '%s'\n", stream_type);
        return CMP_MATCH;
    } else {
        return CMP_STOP;
    }
}
 
void ast_sip_session_unregister_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type)
{
    ao2_callback_data(sdp_handlers, OBJ_KEY | OBJ_UNLINK | OBJ_NODATA, remove_handler, (void *)stream_type, handler);
}
 
static int media_stats_local_ssrc_cmp(
        const struct ast_rtp_instance_stats *vec_elem, const struct ast_rtp_instance_stats *srch)
{
    if (vec_elem->local_ssrc == srch->local_ssrc) {
        return 1;
    }
 
    return 0;
}
 
static struct ast_sip_session_media_state *internal_sip_session_media_state_alloc(
    size_t sessions, size_t read_callbacks)
{
    struct ast_sip_session_media_state *media_state;
 
    media_state = ast_calloc(1, sizeof(*media_state));
    if (!media_state) {
        return NULL;
    }
 
    if (AST_VECTOR_INIT(&media_state->sessions, sessions) < 0) {
        ast_free(media_state);
        return NULL;
    }
 
    if (AST_VECTOR_INIT(&media_state->read_callbacks, read_callbacks) < 0) {
        AST_VECTOR_FREE(&media_state->sessions);
        ast_free(media_state);
        return NULL;
    }
 
    return media_state;
}
 
struct ast_sip_session_media_state *ast_sip_session_media_state_alloc(void)
{
    return internal_sip_session_media_state_alloc(
        DEFAULT_NUM_SESSION_MEDIA, DEFAULT_NUM_SESSION_MEDIA);
}
 
void ast_sip_session_media_stats_save(struct ast_sip_session *sip_session, struct ast_sip_session_media_state *media_state)
{
    int i;
    int ret;
 
    if (!media_state || !sip_session) {
        return;
    }
 
    for (i = 0; i < AST_VECTOR_SIZE(&media_state->sessions); i++) {
        struct ast_rtp_instance_stats *stats_tmp = NULL;
        struct ast_sip_session_media *media = AST_VECTOR_GET(&media_state->sessions, i);
        if (!media || !media->rtp) {
            continue;
        }
 
        stats_tmp = ast_calloc(1, sizeof(struct ast_rtp_instance_stats));
        if (!stats_tmp) {
            return;
        }
 
        ret = ast_rtp_instance_get_stats(media->rtp, stats_tmp, AST_RTP_INSTANCE_STAT_ALL);
        if (ret) {
            ast_free(stats_tmp);
            continue;
        }
 
        /* remove all the duplicated stats if exist */
        AST_VECTOR_REMOVE_CMP_UNORDERED(&sip_session->media_stats, stats_tmp, media_stats_local_ssrc_cmp, ast_free);
 
        AST_VECTOR_APPEND(&sip_session->media_stats, stats_tmp);
    }
}
 
void ast_sip_session_media_state_reset(struct ast_sip_session_media_state *media_state)
{
    int index;
 
    if (!media_state) {
        return;
    }
 
    AST_VECTOR_RESET(&media_state->sessions, ao2_cleanup);
    AST_VECTOR_RESET(&media_state->read_callbacks, AST_VECTOR_ELEM_CLEANUP_NOOP);
 
    for (index = 0; index < AST_MEDIA_TYPE_END; ++index) {
        media_state->default_session[index] = NULL;
    }
 
    ast_stream_topology_free(media_state->topology);
    media_state->topology = NULL;
}
 
struct ast_sip_session_media_state *ast_sip_session_media_state_clone(const struct ast_sip_session_media_state *media_state)
{
    struct ast_sip_session_media_state *cloned;
    int index;
 
    if (!media_state) {
        return NULL;
    }
 
    cloned = internal_sip_session_media_state_alloc(
        AST_VECTOR_SIZE(&media_state->sessions),
        AST_VECTOR_SIZE(&media_state->read_callbacks));
    if (!cloned) {
        return NULL;
    }
 
    if (media_state->topology) {
        cloned->topology = ast_stream_topology_clone(media_state->topology);
        if (!cloned->topology) {
            ast_sip_session_media_state_free(cloned);
            return NULL;
        }
    }
 
    for (index = 0; index < AST_VECTOR_SIZE(&media_state->sessions); ++index) {
        struct ast_sip_session_media *session_media = AST_VECTOR_GET(&media_state->sessions, index);
        enum ast_media_type type = ast_stream_get_type(ast_stream_topology_get_stream(cloned->topology, index));
 
        ao2_bump(session_media);
        if (AST_VECTOR_REPLACE(&cloned->sessions, index, session_media)) {
            ao2_cleanup(session_media);
        }
        if (ast_stream_get_state(ast_stream_topology_get_stream(cloned->topology, index)) != AST_STREAM_STATE_REMOVED &&
            !cloned->default_session[type]) {
            cloned->default_session[type] = session_media;
        }
    }
 
    for (index = 0; index < AST_VECTOR_SIZE(&media_state->read_callbacks); ++index) {
        struct ast_sip_session_media_read_callback_state *read_callback = AST_VECTOR_GET_ADDR(&media_state->read_callbacks, index);
 
        AST_VECTOR_REPLACE(&cloned->read_callbacks, index, *read_callback);
    }
 
    return cloned;
}
 
void ast_sip_session_media_state_free(struct ast_sip_session_media_state *media_state)
{
    if (!media_state) {
        return;
    }
 
    /* This will reset the internal state so we only have to free persistent things */
    ast_sip_session_media_state_reset(media_state);
 
    AST_VECTOR_FREE(&media_state->sessions);
    AST_VECTOR_FREE(&media_state->read_callbacks);
 
    ast_free(media_state);
}
 
int ast_sip_session_is_pending_stream_default(const struct ast_sip_session *session, const struct ast_stream *stream)
{
    int index;
 
    if (!session->pending_media_state->topology) {
        ast_log(LOG_WARNING, "Pending topology was NULL for channel '%s'\n",
            session->channel ? ast_channel_name(session->channel) : "unknown");
        return 0;
    }
 
    if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
        return 0;
    }
 
    for (index = 0; index < ast_stream_topology_get_count(session->pending_media_state->topology); ++index) {
        if (ast_stream_get_type(ast_stream_topology_get_stream(session->pending_media_state->topology, index)) !=
            ast_stream_get_type(stream)) {
            continue;
        }
 
        return ast_stream_topology_get_stream(session->pending_media_state->topology, index) == stream ? 1 : 0;
    }
 
    return 0;
}
 
int ast_sip_session_media_add_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
    int fd, ast_sip_session_media_read_cb callback)
{
    struct ast_sip_session_media_read_callback_state callback_state = {
        .fd = fd,
        .read_callback = callback,
        .session = session_media,
    };
 
    /* The contents of the vector are whole structs and not pointers */
    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);
    ast_free(session_media->remote_label);
    ast_free(session_media->stream_name);
}
 
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;
    struct ast_sip_session_media *current_session_media = NULL;
    SCOPE_ENTER(1, "%s Adding position %d\n", ast_sip_session_get_name(session), position);
 
    /* 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)) {
        current_session_media = AST_VECTOR_GET(&media_state->sessions, position);
        if (current_session_media && current_session_media->type == type) {
            SCOPE_EXIT_RTN_VALUE(current_session_media, "Using existing media_session\n");
        }
    }
 
    /* 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);
            ast_trace(1, "Reusing existing media session\n");
            /*
             * 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);
                    SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't alloc mid\n");
                }
            }
        } else {
            ast_trace(1, "Can't reuse existing media session because the types are different. %s <> %s\n",
                ast_codec_media_type2str(type), ast_codec_media_type2str(session_media->type));
            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;
        }
        ast_trace(1, "Creating new media session\n");
 
        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);
                SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't alloc mid\n");
            }
            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;
        }
    }
 
    ast_free(session_media->stream_name);
    session_media->stream_name = ast_strdup(ast_stream_get_name(ast_stream_topology_get_stream(media_state->topology, position)));
 
    if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {
        ao2_ref(session_media, -1);
 
        SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't replace media_session\n");
    }
 
    ao2_cleanup(current_session_media);
 
    /* 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) {
        ast_trace(1, "Setting media session as default for %s\n", ast_codec_media_type2str(session_media->type));
 
        media_state->default_session[type] = session_media;
    }
 
    SCOPE_EXIT_RTN_VALUE(session_media, "Done\n");
}
 
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;
    ast_free(session_media->remote_label);
    session_media->remote_label = 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 };
        static const pj_str_t STR_label = { "label", 5 };
 
        if (!pj_strcmp(&attr->name, &STR_label)) {
            ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
            session_media->remote_label = ast_strdup(attr_value);
        } else 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(3, "%s: Media count: %d\n", ast_sip_session_get_name(session), sdp->media_count);
 
    if (session->inv_session && session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
        SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to handle incoming SDP. Session has been already disconnected\n",
            ast_sip_session_get_name(session));
    }
 
    /* 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_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc pending topology\n",
                ast_sip_session_get_name(session));
        }
    }
 
    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];
        SCOPE_ENTER(4, "%s: Processing stream %d\n", ast_sip_session_get_name(session), 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);
            ast_trace(-1, "%s: Using existing pending stream %s\n", ast_sip_session_get_name(session),
                ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
        }
        if (!stream) {
            struct ast_stream *existing_stream = NULL;
            char *stream_name = NULL, *stream_name_allocated = NULL;
            const char *stream_label = 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);
                ast_trace(-1, "%s: Found existing active stream %s\n", ast_sip_session_get_name(session),
                    ast_str_tmp(128, ast_stream_to_str(existing_stream, &STR_TMP)));
 
                if (ast_stream_get_state(existing_stream) != AST_STREAM_STATE_REMOVED) {
                    stream_name = (char *)ast_stream_get_name(existing_stream);
                    stream_label = ast_stream_get_metadata(existing_stream, "SDP:LABEL");
                }
            }
 
            if (ast_strlen_zero(stream_name)) {
                if (ast_asprintf(&stream_name_allocated, "%s-%d", ast_codec_media_type2str(type), i) < 0) {
                    handled = 0;
                    SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc stream name\n",
                         ast_sip_session_get_name(session));
 
                }
                stream_name = stream_name_allocated;
                ast_trace(-1, "%s: Using %s for new stream name\n", ast_sip_session_get_name(session),
                    stream_name);
            }
 
            stream = ast_stream_alloc(stream_name, type);
            ast_free(stream_name_allocated);
            if (!stream) {
                handled = 0;
                SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc stream\n",
                     ast_sip_session_get_name(session));
            }
 
            if (!ast_strlen_zero(stream_label)) {
                ast_stream_set_metadata(stream, "SDP:LABEL", stream_label);
                ast_trace(-1, "%s: Using %s for new stream label\n", ast_sip_session_get_name(session),
                    stream_label);
 
            }
 
            if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
                ast_stream_free(stream);
                handled = 0;
                SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't set stream in topology\n",
                     ast_sip_session_get_name(session));
            }
 
            /* 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);
            }
            ast_trace(-1, "%s: Using new stream %s\n", ast_sip_session_get_name(session),
                ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
        }
 
        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_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't alloc session media\n",
                 ast_sip_session_get_name(session));
        }
 
        /* 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)) {
            remove_stream_from_bundle(session_media, stream);
            SCOPE_EXIT_EXPR(continue, "%s: Declining incoming SDP media stream %s'\n",
                ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
        }
 
        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_trace(-1, "%s: Negotiating incoming SDP media stream %s using %s SDP handler\n",
                ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)),
                session_media->handler->id);
            res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
            if (res < 0) {
                /* Catastrophic failure. Abort! */
                SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't negotiate stream %s\n",
                     ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
            } else if (res == 0) {
                remove_stream_from_bundle(session_media, stream);
                SCOPE_EXIT_EXPR(continue, "%s: Declining incoming SDP media stream %s\n",
                    ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
            } else if (res > 0) {
                handled = 1;
                ++type_streams[type];
                /* Handled by this handler. Move to the next stream */
                SCOPE_EXIT_EXPR(continue, "%s: Media stream %s handled by %s\n",
                    ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)),
                    session_media->handler->id);
            }
        }
 
        handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
        if (!handler_list) {
            SCOPE_EXIT_EXPR(continue, "%s: Media stream %s has no registered handlers\n",
                ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
        }
        AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
            if (handler == session_media->handler) {
                continue;
            }
            ast_trace(-1, "%s: Negotiating incoming SDP media stream %s using %s SDP handler\n",
                ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)),
                handler->id);
 
            res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
            if (res < 0) {
                /* Catastrophic failure. Abort! */
                handled = 0;
                SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Couldn't negotiate stream %s\n",
                     ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
            } else if (res == 0) {
                remove_stream_from_bundle(session_media, stream);
                ast_trace(-1, "%s: Declining incoming SDP media stream %s\n",
                    ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
                continue;
            } else if (res > 0) {
                session_media_set_handler(session_media, handler);
                handled = 1;
                ++type_streams[type];
                ast_trace(-1, "%s: Media stream %s handled by %s\n",
                    ast_sip_session_get_name(session), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)),
                    session_media->handler->id);
                break;
            }
        }
 
        SCOPE_EXIT("%s: Done with stream %s\n", ast_sip_session_get_name(session),
            ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
    }
 
end:
    SCOPE_EXIT_RTN_VALUE(handled ? 0 : -1, "%s: Handled? %s\n", ast_sip_session_get_name(session),
        handled ? "yes" : "no");
}
 
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(4, "%s: Applying 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);
        res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
        if (res >= 0) {
            ast_debug(4, "%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(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(4, "%s: No registered SDP handlers for media type '%s'\n", ast_sip_session_get_name(session), media);
        return -1;
    }
    AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
        if (handler == session_media->handler) {
            continue;
        }
        ast_debug(4, "%s: Applying 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);
        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(4, "%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);
            /* 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) {
        ast_debug(4, "%s: Stopping SDP media stream '%s' as it is not currently negotiated\n",
            ast_sip_session_get_name(session), 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, "%s: Unable to clone active media state\n",
                    ast_sip_session_get_name(session));
                return -1;
            }
 
            ast_sip_session_media_state_free(session->pending_media_state);
            session->pending_media_state = active_media_state_clone;
        } else {
            ast_log(LOG_WARNING, "%s: No pending or active media state\n",
                ast_sip_session_get_name(session));
            return -1;
        }
    }
 
    /* 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, "%s: Local SDP contains %d media streams while we expected it to contain %u\n",
            ast_sip_session_get_name(session),
            ast_stream_topology_get_count(session->pending_media_state->topology), local->media_count);
        SCOPE_EXIT_RTN_VALUE(-1, "Media stream count mismatch\n");
    }
 
    AST_VECTOR_RESET(&session->pending_media_state->read_callbacks, AST_VECTOR_ELEM_CLEANUP_NOOP);
 
    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 *pending_media_state;
    /*! Active media state at the time of the original request */
    struct ast_sip_session_media_state *active_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 *pending_media_state,
    struct ast_sip_session_media_state *active_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->pending_media_state = pending_media_state;
    delay->active_media_state = active_media_state;
    return delay;
}
 
static void delayed_request_free(struct ast_sip_session_delayed_request *delay)
{
    ast_sip_session_media_state_free(delay->pending_media_state);
    ast_sip_session_media_state_free(delay->active_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;
    SCOPE_ENTER(3, "%s: sending delayed %s request\n",
        ast_sip_session_get_name(session),
        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->pending_media_state,
            delay->active_media_state, 1);
        /* Ownership of media state transitions to ast_sip_session_refresh */
        delay->pending_media_state = NULL;
        delay->active_media_state = NULL;
        SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_sip_session_get_name(session));
    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->pending_media_state,
            delay->active_media_state, 1);
        /* Ownership of media state transitions to ast_sip_session_refresh */
        delay->pending_media_state = NULL;
        delay->active_media_state = NULL;
        SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_sip_session_get_name(session));
    case DELAYED_METHOD_BYE:
        ast_sip_session_terminate(session, 0);
        SCOPE_EXIT_RTN_VALUE(0, "%s: Terminating session on delayed BYE\n", ast_sip_session_get_name(session));
    }
 
    SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Don't know how to send delayed %s(%d) request.\n",
        ast_sip_session_get_name(session),
        delayed_method2str(delay->method), delay->method);
}
 
/*!
 * \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;
    SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
    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);
            ast_trace(-1, "%s: Sending delayed %s request\n", ast_sip_session_get_name(session),
                delayed_method2str(delay->method));
            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);
    SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_sip_session_get_name(session));
}
 
/*!
 * \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;
    SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
    /* 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);
            ast_trace(-1, "%s: Sending delayed %s request\n", ast_sip_session_get_name(session),
                delayed_method2str(delay->method));
            res = send_delayed_request(session, delay);
            delayed_request_free(delay);
            if (!res) {
                break;
            }
        }
    }
    AST_LIST_TRAVERSE_SAFE_END;
 
    ao2_ref(session, -1);
    SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_sip_session_get_name(session));
}
 
/*!
 * \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;
    SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
    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);
    }
 
    SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_sip_session_get_name(session));
}
 
/*!
 * \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 *pending_media_state,
    struct ast_sip_session_media_state *active_media_state,
    int queue_head)
{
    struct ast_sip_session_delayed_request *delay = delayed_request_alloc(method,
            on_request, on_sdp_creation, on_response, generate_new_sdp, pending_media_state,
            active_media_state);
    SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
    if (!delay) {
        ast_sip_session_media_state_free(pending_media_state);
        ast_sip_session_media_state_free(active_media_state);
        SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "Unable to allocate delay request\n");
    }
 
    if (method == DELAYED_METHOD_BYE || queue_head) {
        /* 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);
    }
    SCOPE_EXIT_RTN_VALUE(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(3, "%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, 0));
}
 
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, "%s: From header domain reset by channel variable SIPFROMDOMAIN (%s)\n",
            ast_sip_session_get_name(session), 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);
    }
}
 
/*
 * Helper macros for merging and validating media states
 */
#define STREAM_REMOVED(_stream) (ast_stream_get_state(_stream) == AST_STREAM_STATE_REMOVED)
#define STATE_REMOVED(_stream_state) (_stream_state == AST_STREAM_STATE_REMOVED)
#define STATE_NONE(_stream_state) (_stream_state == AST_STREAM_STATE_END)
#define GET_STREAM_SAFE(_topology, _i) (_i < ast_stream_topology_get_count(_topology) ? ast_stream_topology_get_stream(_topology, _i) : NULL)
#define GET_STREAM_STATE_SAFE(_stream) (_stream ? ast_stream_get_state(_stream) : AST_STREAM_STATE_END)
#define GET_STREAM_NAME_SAFE(_stream) (_stream ? ast_stream_get_name(_stream) : "")
 
/*!
 * \internal
 * \brief Validate a media state
 *
 * \param session_name For log messages
 * \param state Media state
 *
 * \retval 1 The media state is valid
 * \retval 0 The media state is NOT valid
 *
 */
static int is_media_state_valid(const char *session_name, struct ast_sip_session_media_state *state)
{
    int stream_count = ast_stream_topology_get_count(state->topology);
    int session_count = AST_VECTOR_SIZE(&state->sessions);
    int i;
    int res = 0;
    SCOPE_ENTER(3, "%s: Topology: %s\n", session_name,
        ast_str_tmp(256, ast_stream_topology_to_str(state->topology, &STR_TMP)));
 
    if (session_count != stream_count) {
        SCOPE_EXIT_RTN_VALUE(0, "%s: %d media sessions but %d streams\n", session_name,
            session_count, stream_count);
    }
 
    for (i = 0; i < stream_count; i++) {
        struct ast_sip_session_media *media = NULL;
        struct ast_stream *stream = ast_stream_topology_get_stream(state->topology, i);
        const char *stream_name = NULL;
        int j;
        SCOPE_ENTER(4, "%s: Checking stream %s\n", session_name, ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
 
        if (!stream) {
            SCOPE_EXIT_EXPR(goto end, "%s: stream %d is null\n", session_name, i);
        }
        stream_name = ast_stream_get_name(stream);
 
        for (j = 0; j < stream_count; j++) {
            struct ast_stream *possible_dup = ast_stream_topology_get_stream(state->topology, j);
            if (j == i || !possible_dup) {
                continue;
            }
            if (!STREAM_REMOVED(stream) && ast_strings_equal(stream_name, GET_STREAM_NAME_SAFE(possible_dup))) {
                SCOPE_EXIT_EXPR(goto end, "%s: stream %i %s is duplicated to %d\n", session_name,
                    i, stream_name, j);
            }
        }
 
        media = AST_VECTOR_GET(&state->sessions, i);
        if (!media) {
            SCOPE_EXIT_EXPR(continue, "%s: media %d is null\n", session_name, i);
        }
 
        for (j = 0; j < session_count; j++) {
            struct ast_sip_session_media *possible_dup = AST_VECTOR_GET(&state->sessions, j);
            if (j == i || !possible_dup) {
                continue;
            }
            if (!ast_strlen_zero(media->label) && !ast_strlen_zero(possible_dup->label)
                && ast_strings_equal(media->label, possible_dup->label)) {
                SCOPE_EXIT_EXPR(goto end, "%s: media %d %s is duplicated to %d\n", session_name,
                    i, media->label, j);
            }
        }
 
        if (media->stream_num != i) {
            SCOPE_EXIT_EXPR(goto end, "%s: media %d has stream_num %d\n", session_name,
                i, media->stream_num);
        }
 
        if (media->type != ast_stream_get_type(stream)) {
            SCOPE_EXIT_EXPR(goto end, "%s: media %d has type %s but stream has type %s\n", stream_name,
                i, ast_codec_media_type2str(media->type), ast_codec_media_type2str(ast_stream_get_type(stream)));
        }
        SCOPE_EXIT("%s: Done with stream %s\n", session_name, ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
    }
 
    res = 1;
end:
    SCOPE_EXIT_RTN_VALUE(res, "%s: %s\n", session_name, res ? "Valid" : "NOT Valid");
}
 
/*!
 * \internal
 * \brief Merge media states for a delayed session refresh
 *
 * \param session_name For log messages
 * \param delayed_pending_state The pending media state at the time the resuest was queued
 * \param delayed_active_state The active media state  at the time the resuest was queued
 * \param current_active_state The current active media state
 * \param run_post_validation Whether to run validation on the resulting media state or not
 *
 * \returns New merged topology or NULL if there's an error
 *
 */
static struct ast_sip_session_media_state *resolve_refresh_media_states(
    const char *session_name,
    struct ast_sip_session_media_state *delayed_pending_state,
    struct ast_sip_session_media_state *delayed_active_state,
    struct ast_sip_session_media_state *current_active_state,
    int run_post_validation)
{
    RAII_VAR(struct ast_sip_session_media_state *, new_pending_state, NULL, ast_sip_session_media_state_free);
    struct ast_sip_session_media_state *returned_media_state = NULL;
    struct ast_stream_topology *delayed_pending = delayed_pending_state->topology;
    struct ast_stream_topology *delayed_active = delayed_active_state->topology;
    struct ast_stream_topology *current_active = current_active_state->topology;
    struct ast_stream_topology *new_pending = NULL;
    int i;
    int max_stream_count;
    int res;
    SCOPE_ENTER(2, "%s: DP: %s  DA: %s  CA: %s\n", session_name,
        ast_str_tmp(256, ast_stream_topology_to_str(delayed_pending, &STR_TMP)),
        ast_str_tmp(256, ast_stream_topology_to_str(delayed_active, &STR_TMP)),
        ast_str_tmp(256, ast_stream_topology_to_str(current_active, &STR_TMP))
    );
 
    max_stream_count = MAX(ast_stream_topology_get_count(delayed_pending),
        ast_stream_topology_get_count(delayed_active));
    max_stream_count = MAX(max_stream_count, ast_stream_topology_get_count(current_active));
 
    /*
     * The new_pending_state is always based on the currently negotiated state because
     * the stream ordering in its topology must be preserved.
     */
    new_pending_state = ast_sip_session_media_state_clone(current_active_state);
    if (!new_pending_state) {
        SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "%s: Couldn't clone current_active_state to new_pending_state\n", session_name);
    }
    new_pending = new_pending_state->topology;
 
    for (i = 0; i < max_stream_count; i++) {
        struct ast_stream *dp_stream = GET_STREAM_SAFE(delayed_pending, i);
        struct ast_stream *da_stream = GET_STREAM_SAFE(delayed_active, i);
        struct ast_stream *ca_stream = GET_STREAM_SAFE(current_active, i);
        struct ast_stream *np_stream = GET_STREAM_SAFE(new_pending, i);
        struct ast_stream *found_da_stream = NULL;
        struct ast_stream *found_np_stream = NULL;
        enum ast_stream_state dp_state = GET_STREAM_STATE_SAFE(dp_stream);
        enum ast_stream_state da_state = GET_STREAM_STATE_SAFE(da_stream);
        enum ast_stream_state ca_state = GET_STREAM_STATE_SAFE(ca_stream);
        enum ast_stream_state np_state = GET_STREAM_STATE_SAFE(np_stream);
        enum ast_stream_state found_da_state = AST_STREAM_STATE_END;
        enum ast_stream_state found_np_state = AST_STREAM_STATE_END;
        const char *da_name = GET_STREAM_NAME_SAFE(da_stream);
        const char *dp_name = GET_STREAM_NAME_SAFE(dp_stream);
        const char *ca_name = GET_STREAM_NAME_SAFE(ca_stream);
        const char *np_name = GET_STREAM_NAME_SAFE(np_stream);
        const char *found_da_name __attribute__((unused)) = "";
        const char *found_np_name __attribute__((unused)) = "";
        int found_da_slot __attribute__((unused)) = -1;
        int found_np_slot = -1;
        int removed_np_slot = -1;
        int j;
        SCOPE_ENTER(3, "%s: slot: %d DP: %s  DA: %s  CA: %s\n", session_name, i,
            ast_str_tmp(128, ast_stream_to_str(dp_stream, &STR_TMP)),
            ast_str_tmp(128, ast_stream_to_str(da_stream, &STR_TMP)),
            ast_str_tmp(128, ast_stream_to_str(ca_stream, &STR_TMP)));
 
        if (STATE_NONE(da_state) && STATE_NONE(dp_state) && STATE_NONE(ca_state)) {
            SCOPE_EXIT_EXPR(break, "%s: All gone\n", session_name);
        }
 
        /*
         * Simple cases are handled first to avoid having to search the NP and DA
         * topologies for streams with the same name but not in the same position.
         */
 
        if (STATE_NONE(dp_state) && !STATE_NONE(da_state)) {
            /*
             * The slot in the delayed pending topology can't be empty if the delayed
             * active topology has a stream there.  Streams can't just go away.  They
             * can be reused or marked "removed" but they can't go away.
             */
            SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: DP slot is empty but DA is not\n", session_name);
        }
 
        if (STATE_NONE(dp_state)) {
            /*
             * The current active topology can certainly have streams that weren't
             * in existence when the delayed request was queued.  In this case,
             * no action is needed since we already copied the current active topology
             * to the new pending one.
             */
            SCOPE_EXIT_EXPR(continue, "%s: No DP stream so use CA stream as is\n", session_name);
        }
 
        if (ast_strings_equal(dp_name, da_name) && ast_strings_equal(da_name, ca_name)) {
            /*
             * The delayed pending stream in this slot matches by name, the streams
             * in the same slot in the other two topologies.  Easy case.
             */
            ast_trace(-1, "%s: Same stream in all 3 states\n", session_name);
            if (dp_state == da_state && da_state == ca_state) {
                /* All the same state, no need to update. */
                SCOPE_EXIT_EXPR(continue, "%s: All in the same state so nothing to do\n", session_name);
            }
            if (da_state != ca_state) {
                /*
                 * Something set the CA state between the time this request was queued
                 * and now.  The CA state wins so we don't do anything.
                 */
                SCOPE_EXIT_EXPR(continue, "%s: Ignoring request to change state from %s to %s\n",
                    session_name, ast_stream_state2str(ca_state), ast_stream_state2str(dp_state));
            }
            if (dp_state != da_state) {
                /* DP needs to update the state */
                ast_stream_set_state(np_stream, dp_state);
                SCOPE_EXIT_EXPR(continue, "%s: Changed NP stream state from %s to %s\n",
                    session_name, ast_stream_state2str(ca_state), ast_stream_state2str(dp_state));
            }
        }
 
        /*
         * We're done with the simple cases.  For the rest, we need to identify if the
         * DP stream we're trying to take action on is already in the other topologies
         * possibly in a different slot.  To do that, if the stream in the DA or CA slots
         * doesn't match the current DP stream, we need to iterate over the topology
         * looking for a stream with the same name.
         */
 
        /*
         * Since we already copied all of the CA streams to the NP topology, we'll use it
         * instead of CA because we'll be updating the NP as we go.
         */
        if (!ast_strings_equal(dp_name, np_name)) {
            /*
             * The NP stream in this slot doesn't have the same name as the DP stream
             * so we need to see if it's in another NP slot.  We're not going to stop
             * when we find a matching stream because we also want to find the first
             * removed removed slot, if any, so we can re-use this slot.  We'll break
             * early if we find both before we reach the end.
             */
            ast_trace(-1, "%s: Checking if DP is already in NP somewhere\n", session_name);
            for (j = 0; j < ast_stream_topology_get_count(new_pending); j++) {
                struct ast_stream *possible_existing = ast_stream_topology_get_stream(new_pending, j);
                const char *possible_existing_name = GET_STREAM_NAME_SAFE(possible_existing);
 
                ast_trace(-1, "%s: Checking %s against %s\n", session_name, dp_name, possible_existing_name);
                if (found_np_slot == -1 && ast_strings_equal(dp_name, possible_existing_name)) {
                    ast_trace(-1, "%s: Pending stream %s slot %d is in NP slot %d\n", session_name,
                    dp_name, i, j);
                    found_np_slot = j;
                    found_np_stream = possible_existing;
                    found_np_state = ast_stream_get_state(possible_existing);
                    found_np_name = ast_stream_get_name(possible_existing);
                }
                if (STREAM_REMOVED(possible_existing) && removed_np_slot == -1) {
                    removed_np_slot = j;
                }
                if (removed_np_slot >= 0 && found_np_slot >= 0) {
                    break;
                }
            }
        } else {
            /* Makes the subsequent code easier */
            found_np_slot = i;
            found_np_stream = np_stream;
            found_np_state = np_state;
            found_np_name = np_name;
        }
 
        if (!ast_strings_equal(dp_name, da_name)) {
            /*
             * The DA stream in this slot doesn't have the same name as the DP stream
             * so we need to see if it's in another DA slot.  In real life, the DA stream
             * in this slot could have a different name but there shouldn't be a case
             * where the DP stream is another slot in the DA topology.  Just in case though.
             * We don't care about removed slots in the DA topology.
             */
            ast_trace(-1, "%s: Checking if DP is already in DA somewhere\n", session_name);
            for (j = 0; j < ast_stream_topology_get_count(delayed_active); j++) {
                struct ast_stream *possible_existing = ast_stream_topology_get_stream(delayed_active, j);
                const char *possible_existing_name = GET_STREAM_NAME_SAFE(possible_existing);
 
                ast_trace(-1, "%s: Checking %s against %s\n", session_name, dp_name, possible_existing_name);
                if (ast_strings_equal(dp_name, possible_existing_name)) {
                    ast_trace(-1, "%s: Pending stream %s slot %d is already in delayed active slot %d\n",
                        session_name, dp_name, i, j);
                    found_da_slot = j;
                    found_da_stream = possible_existing;
                    found_da_state = ast_stream_get_state(possible_existing);
                    found_da_name = ast_stream_get_name(possible_existing);
                    break;
                }
            }
        } else {
            /* Makes the subsequent code easier */
            found_da_slot = i;
            found_da_stream = da_stream;
            found_da_state = da_state;
            found_da_name = da_name;
        }
 
        ast_trace(-1, "%s: Found NP slot: %d  Found removed NP slot: %d Found DA slot: %d\n",
            session_name, found_np_slot, removed_np_slot, found_da_slot);
 
        /*
         * Now we know whether the DP stream is new or changing state and we know if the DP
         * stream exists in the other topologies and if so, where in those topologies it exists.
         */
 
        if (!found_da_stream) {
            /*
             * The DP stream isn't in the DA topology which would imply that the intention of the
             * request was to add the stream, not change its state.  It's possible though that
             * the stream was added by another request between the time this request was queued
             * and now so we need to check the CA topology as well.
             */
            ast_trace(-1, "%s: There was no corresponding DA stream so the request was to add a stream\n", session_name);
 
            if (found_np_stream) {
                /*
                 * We found it in the CA topology.  Since the intention was to add it
                 * and it's already there, there's nothing to do.
                 */
                SCOPE_EXIT_EXPR(continue, "%s: New stream requested but it's already in CA\n", session_name);
            } else {
                /* OK, it's not in either which would again imply that the intention of the
                 * request was to add the stream.
                 */
                ast_trace(-1, "%s: There was no corresponding NP stream\n", session_name);
                if (STATE_REMOVED(dp_state)) {
                    /*
                     * How can DP request to remove a stream that doesn't seem to exist anythere?
                     * It's not.  It's possible that the stream was already removed and the slot
                     * reused in the CA topology, but it would still have to exist in the DA
                     * topology.  Bail.
                     */
                    SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR,
                        "%s: Attempting to remove stream %d:%s but it doesn't exist anywhere.\n", session_name, i, dp_name);
                } else {
                    /*
                     * We're now sure we want to add the the stream.  Since we can re-use
                     * slots in the CA topology that have streams marked as "removed", we
                     * use the slot we saved in removed_np_slot if it exists.
                     */
                    ast_trace(-1, "%s: Checking for open slot\n", session_name);
                    if (removed_np_slot >= 0) {
                        struct ast_sip_session_media *old_media = AST_VECTOR_GET(&new_pending_state->sessions, removed_np_slot);
                        res = ast_stream_topology_set_stream(new_pending, removed_np_slot, ast_stream_clone(dp_stream, NULL));
                        if (res != 0) {
                            SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't set stream in new topology\n", session_name);
                        }
                        /*
                         * Since we're reusing the removed_np_slot slot for something else, we need
                         * to free and remove any session media already in it.
                         * ast_stream_topology_set_stream() took care of freeing the old stream.
                         */
                        res = AST_VECTOR_REPLACE(&new_pending_state->sessions, removed_np_slot, NULL);
                        if (res != 0) {
                            SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't replace media session\n", session_name);
                        }
 
                        ao2_cleanup(old_media);
                        SCOPE_EXIT_EXPR(continue, "%s: Replaced removed stream in slot %d\n",
                            session_name, removed_np_slot);
                    } else {
                        int new_slot = ast_stream_topology_append_stream(new_pending, ast_stream_clone(dp_stream, NULL));
                        if (new_slot < 0) {
                            SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't append stream in new topology\n", session_name);
                        }
 
                        res = AST_VECTOR_REPLACE(&new_pending_state->sessions, new_slot, NULL);
                        if (res != 0) {
                            SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_WARNING, "%s: Couldn't replace media session\n", session_name);
                        }
                        SCOPE_EXIT_EXPR(continue, "%s: Appended new stream to slot %d\n",
                            session_name, new_slot);
                    }
                }
            }
        } else {
            /*
             * The DP stream exists in the DA topology so it's a change of some sort.
             */
            ast_trace(-1, "%s: There was a corresponding DA stream so the request was to change/remove a stream\n", session_name);
            if (dp_state == found_da_state) {
                /* No change? Let's see if it's in CA */
                if (!found_np_stream) {
                    /*
                     * The DP and DA state are the same which would imply that the stream
                     * already exists but it's not in the CA topology.  It's possible that
                     * between the time this request was queued and now the stream was removed
                     * from the CA topology and the slot used for something else.  Nothing
                     * we can do here.
                     */
                    SCOPE_EXIT_EXPR(continue, "%s: Stream doesn't exist in CA so nothing to do\n", session_name);
                } else if (dp_state == found_np_state) {
                    SCOPE_EXIT_EXPR(continue, "%s: States are the same all around so nothing to do\n", session_name);
                } else {
                    SCOPE_EXIT_EXPR(continue, "%s: Something changed the CA state so we're going to leave it as is\n", session_name);
                }
            } else {
                /* We have a state change. */
                ast_trace(-1, "%s: Requesting state change to %s\n", session_name, ast_stream_state2str(dp_state));
                if (!found_np_stream) {
                    SCOPE_EXIT_EXPR(continue, "%s: Stream doesn't exist in CA so nothing to do\n", session_name);
                } else if (da_state == found_np_state) {
                    ast_stream_set_state(found_np_stream, dp_state);
                    SCOPE_EXIT_EXPR(continue, "%s: Changed NP stream state from %s to %s\n",
                        session_name, ast_stream_state2str(found_np_state), ast_stream_state2str(dp_state));
                } else {
                    SCOPE_EXIT_EXPR(continue, "%s: Something changed the CA state so we're going to leave it as is\n",
                        session_name);
                }
            }
        }
 
        SCOPE_EXIT("%s: Done with slot %d\n", session_name, i);
    }
 
    ast_trace(-1, "%s: Resetting default media states\n", session_name);
    for (i = 0; i < AST_MEDIA_TYPE_END; i++) {
        int j;
        new_pending_state->default_session[i] = NULL;
        for (j = 0; j < AST_VECTOR_SIZE(&new_pending_state->sessions); j++) {
            struct ast_sip_session_media *media = AST_VECTOR_GET(&new_pending_state->sessions, j);
            struct ast_stream *stream = ast_stream_topology_get_stream(new_pending_state->topology, j);
 
            if (media && media->type == i && !STREAM_REMOVED(stream)) {
                new_pending_state->default_session[i] = media;
                break;
            }
        }
    }
 
    if (run_post_validation) {
        ast_trace(-1, "%s: Running post-validation\n", session_name);
        if (!is_media_state_valid(session_name, new_pending_state)) {
            SCOPE_EXIT_LOG_RTN_VALUE(NULL, LOG_ERROR, "State not consistent\n");
        }
    }
 
    /*
     * We need to move the new pending state to another variable and set new_pending_state to NULL
     * so RAII_VAR doesn't free it.
     */
    returned_media_state = new_pending_state;
    new_pending_state = NULL;
    SCOPE_EXIT_RTN_VALUE(returned_media_state, "%s: NP: %s\n", session_name,
        ast_str_tmp(256, ast_stream_topology_to_str(new_pending, &STR_TMP)));
}
 
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 *pending_media_state,
        struct ast_sip_session_media_state *active_media_state,
        int queued)
{
    pjsip_inv_session *inv_session = session->inv_session;
    pjmedia_sdp_session *new_sdp = NULL;
    pjsip_tx_data *tdata;
    int res = -1;
    SCOPE_ENTER(3, "%s: New SDP? %s  Queued? %s DP: %s  DA: %s\n", ast_sip_session_get_name(session),
        generate_new_sdp ? "yes" : "no", queued ? "yes" : "no",
        pending_media_state ? ast_str_tmp(256, ast_stream_topology_to_str(pending_media_state->topology, &STR_TMP)) : "none",
        active_media_state ? ast_str_tmp(256, ast_stream_topology_to_str(active_media_state->topology, &STR_TMP)) : "none");
 
    if (pending_media_state && (!pending_media_state->topology || !generate_new_sdp)) {
 
        ast_sip_session_media_state_free(pending_media_state);
        ast_sip_session_media_state_free(active_media_state);
        SCOPE_EXIT_RTN_VALUE(-1, "%s: Not sending reinvite because %s%s\n", ast_sip_session_get_name(session),
            pending_media_state->topology == NULL ? "pending topology is null " : "",
                !generate_new_sdp ? "generate_new_sdp is false" : "");
    }
 
    if (inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
        /* Don't try to do anything with a hung-up call */
        ast_sip_session_media_state_free(pending_media_state);
        ast_sip_session_media_state_free(active_media_state);
        SCOPE_EXIT_RTN_VALUE(0, "%s: Not sending reinvite because of disconnected state\n",
                ast_sip_session_get_name(session));
    }
 
    /* If the dialog has not yet been established we have to defer until it has */
    if (inv_session->dlg->state != PJSIP_DIALOG_STATE_ESTABLISHED) {
        res = 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,
            pending_media_state, active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
        SCOPE_EXIT_RTN_VALUE(res, "%s: Delay sending reinvite because dialog has not been established\n",
            ast_sip_session_get_name(session));
    }
 
    if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
        if (inv_session->invite_tsx) {
            /* We can't send a reinvite yet, so delay it */
            res = delay_request(session, on_request_creation, on_sdp_creation,
                on_response, generate_new_sdp, DELAYED_METHOD_INVITE, pending_media_state,
                active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
            SCOPE_EXIT_RTN_VALUE(res, "%s: Delay sending reinvite because of outstanding transaction\n",
                ast_sip_session_get_name(session));
        } 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_sip_session_media_state_free(pending_media_state);
            ast_sip_session_media_state_free(active_media_state);
            SCOPE_EXIT_RTN_VALUE(0, "%s: Not sending reinvite because not in confirmed state\n",
                ast_sip_session_get_name(session));
        }
    }
 
    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) {
            res = 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, pending_media_state,
                active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
            SCOPE_EXIT_RTN_VALUE(res, "%s: Delay session refresh with new SDP because SDP negotiation is not yet done\n",
                ast_sip_session_get_name(session));
        }
 
        /* If an explicitly requested media state has been provided use it instead of any pending one */
        if (pending_media_state) {
            int index;
            int type_streams[AST_MEDIA_TYPE_END] = {0};
 
            ast_trace(-1, "%s: Pending media state exists\n", ast_sip_session_get_name(session));
 
            /* 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)) {
                res = 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, pending_media_state,
                        active_media_state ? active_media_state : ast_sip_session_media_state_clone(session->active_media_state), queued);
                SCOPE_EXIT_RTN_VALUE(res, "%s: Delay sending reinvite because of outstanding requests\n",
                    ast_sip_session_get_name(session));
            }
 
            /*
             * Attempt to resolve only if objects are available, and it's not
             * switching to or from an image type.
             */
            if (active_media_state && active_media_state->topology &&
                (!active_media_state->default_session[AST_MEDIA_TYPE_IMAGE] ==
                 !pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE])) {
 
                struct ast_sip_session_media_state *new_pending_state;
 
                ast_trace(-1, "%s: Active media state exists and is%s equal to pending\n", ast_sip_session_get_name(session),
                    !ast_stream_topology_equal(active_media_state->topology,pending_media_state->topology) ? " not" : "");
                ast_trace(-1, "%s: DP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(pending_media_state->topology, &STR_TMP)));
                ast_trace(-1, "%s: DA: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(active_media_state->topology, &STR_TMP)));
                ast_trace(-1, "%s: CP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)));
                ast_trace(-1, "%s: CA: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
 
                new_pending_state = resolve_refresh_media_states(ast_sip_session_get_name(session),
                    pending_media_state, active_media_state, session->active_media_state, 1);
                if (new_pending_state) {
                    ast_trace(-1, "%s: NP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(new_pending_state->topology, &STR_TMP)));
                    ast_sip_session_media_state_free(pending_media_state);
                    pending_media_state = new_pending_state;
                } else {
                    ast_sip_session_media_state_reset(pending_media_state);
                    ast_sip_session_media_state_free(active_media_state);
                    SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Unable to merge media states\n", ast_sip_session_get_name(session));
                }
            }
 
            /* 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.
             */
            ast_trace(-1, "%s: Pruning and checking formats of streams\n", ast_sip_session_get_name(session));
 
            for (index = 0; index < ast_stream_topology_get_count(pending_media_state->topology); ++index) {
                struct ast_stream *existing_stream = NULL;
                struct ast_stream *stream = ast_stream_topology_get_stream(pending_media_state->topology, index);
                SCOPE_ENTER(4, "%s: Checking stream %s\n", ast_sip_session_get_name(session),
                    ast_stream_get_name(stream));
 
                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);
                    ast_trace(-1, "%s: Found existing stream %s\n", ast_sip_session_get_name(session),
                        ast_stream_get_name(existing_stream));
                }
 
                if (is_stream_limitation_reached(ast_stream_get_type(stream), session->endpoint, type_streams)) {
                    if (index < AST_VECTOR_SIZE(&pending_media_state->sessions)) {
                        struct ast_sip_session_media *session_media = AST_VECTOR_GET(&pending_media_state->sessions, index);
 
                        ao2_cleanup(session_media);
                        AST_VECTOR_REMOVE(&pending_media_state->sessions, index, 1);
                    }
 
                    ast_stream_topology_del_stream(pending_media_state->topology, index);
                    ast_trace(-1, "%s: Dropped overlimit stream %s\n", ast_sip_session_get_name(session),
                        ast_stream_get_name(stream));
 
                    /* A stream has potentially moved into our spot so we need to jump back so we process it */
                    index -= 1;
                    SCOPE_EXIT_EXPR(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_trace(-1, "%s: Dropped removed stream %s\n", ast_sip_session_get_name(session),
                            ast_stream_get_name(stream));
                        ast_stream_topology_del_stream(pending_media_state->topology, index);
                        /* TODO: Do we need to remove the corresponding media state? */
                        index -= 1;
                    }
                    SCOPE_EXIT_EXPR(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(pending_media_state);
                        ast_sip_session_media_state_free(active_media_state);
                        res = -1;
                        SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Unable to alloc format caps\n", ast_sip_session_get_name(session));
                    }
                    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(pending_media_state->topology, index);
                            index -= 1;
                            SCOPE_EXIT_EXPR(continue, "%s: Dropped incompatible stream %s\n",
                                ast_sip_session_get_name(session), ast_stream_get_name(stream));
                        } 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);
                            SCOPE_EXIT_EXPR(continue, "%s: Dropped incompatible stream %s\n",
                                ast_sip_session_get_name(session), ast_stream_get_name(stream));
                        } 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)];
 
                SCOPE_EXIT();
            }
 
            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(pending_media_state->topology);
                    index < ast_stream_topology_get_count(session->active_media_state->topology); ++index) {
                    struct ast_stream *stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
                    struct ast_stream *cloned;
                    int position;
                    SCOPE_ENTER(4, "%s: Stream %s not in pending\n", ast_sip_session_get_name(session),
                        ast_stream_get_name(stream));
 
                    cloned = ast_stream_clone(stream, NULL);
                    if (!cloned) {
                        ast_sip_session_media_state_free(pending_media_state);
                        ast_sip_session_media_state_free(active_media_state);
                        res = -1;
                        SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Unable to clone stream %s\n",
                            ast_sip_session_get_name(session), ast_stream_get_name(stream));
                    }
 
                    ast_stream_set_state(cloned, AST_STREAM_STATE_REMOVED);
                    position = ast_stream_topology_append_stream(pending_media_state->topology, cloned);
                    if (position < 0) {
                        ast_stream_free(cloned);
                        ast_sip_session_media_state_free(pending_media_state);
                        ast_sip_session_media_state_free(active_media_state);
                        res = -1;
                        SCOPE_EXIT_LOG_EXPR(goto end, LOG_ERROR, "%s: Unable to append cloned stream\n",
                            ast_sip_session_get_name(session));
                    }
                    SCOPE_EXIT("%s: Appended empty stream in position %d to make counts match\n",
                        ast_sip_session_get_name(session), position);
                }
 
                /*
                 * We can suppress this re-invite if the pending topology is equal to the currently
                 * active topology.
                 */
                if (ast_stream_topology_equal(session->active_media_state->topology, pending_media_state->topology)) {
                    ast_trace(-1, "%s: CA: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
                    ast_trace(-1, "%s: NP: %s\n", ast_sip_session_get_name(session), ast_str_tmp(256, ast_stream_topology_to_str(pending_media_state->topology, &STR_TMP)));
                    ast_sip_session_media_state_free(pending_media_state);
                    ast_sip_session_media_state_free(active_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
                     */
                    SCOPE_EXIT_RTN_VALUE(queued ? 1 : 0, "%s: Topologies are equal. Not sending re-invite\n",
                        ast_sip_session_get_name(session));
                }
            }
 
            ast_sip_session_media_state_free(session->pending_media_state);
            session->pending_media_state = pending_media_state;
        }
 
        new_sdp = generate_session_refresh_sdp(session);
        if (!new_sdp) {
            ast_sip_session_media_state_reset(session->pending_media_state);
            ast_sip_session_media_state_free(active_media_state);
            SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed to generate session refresh SDP. Not sending session refresh\n",
                ast_sip_session_get_name(session));
        }
        if (on_sdp_creation) {
            if (on_sdp_creation(session, new_sdp)) {
                ast_sip_session_media_state_reset(session->pending_media_state);
                ast_sip_session_media_state_free(active_media_state);
                SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: on_sdp_creation failed\n", ast_sip_session_get_name(session));
            }
        }
    }
 
    if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
        if (pjsip_inv_reinvite(inv_session, NULL, new_sdp, &tdata)) {
            if (generate_new_sdp) {
                ast_sip_session_media_state_reset(session->pending_media_state);
            }
            ast_sip_session_media_state_free(active_media_state);
            SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed to create reinvite properly\n", ast_sip_session_get_name(session));
        }
    } else if (pjsip_inv_update(inv_session, NULL, new_sdp, &tdata)) {
        if (generate_new_sdp) {
            ast_sip_session_media_state_reset(session->pending_media_state);
        }
        ast_sip_session_media_state_free(active_media_state);
        SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: Failed to create UPDATE properly\n", ast_sip_session_get_name(session));
    }
    if (on_request_creation) {
        if (on_request_creation(session, tdata)) {
            if (generate_new_sdp) {
                ast_sip_session_media_state_reset(session->pending_media_state);
            }
            ast_sip_session_media_state_free(active_media_state);
            SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_WARNING, "%s: on_request_creation failed.\n", ast_sip_session_get_name(session));
        }
    }
    ast_sip_session_send_request_with_cb(session, tdata, on_response);
    ast_sip_session_media_state_free(active_media_state);
 
end:
    SCOPE_EXIT_RTN_VALUE(res, "%s: Sending session refresh SDP via %s\n", ast_sip_session_get_name(session),
        method == AST_SIP_SESSION_REFRESH_METHOD_INVITE ? "re-INVITE" : "UPDATE");
}
 
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, NULL, 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, 0);
    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)
{
    pjsip_dialog *dlg = pjsip_tdata_get_dlg(tdata);
    RAII_VAR(struct ast_sip_session *, dlg_session, dlg ? ast_sip_dialog_get_session(dlg) : NULL, ao2_cleanup);
    if (!dlg_session) {
        /* If the dialog has a session, handle_outgoing_response will be called
           from session_on_tx_response. If it does not, call it from here. */
        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 pj_status_t session_on_tx_response(pjsip_tx_data *tdata);
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,
    .on_tx_response = session_on_tx_response,
};
 
/*! \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(3, "%s: No registered SDP handlers for media type '%s'\n", ast_sip_session_get_name(session), 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->inv_session->invite_tsx) {
        /* There's a transaction in progress so bail now and let pjproject send 491 */
        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) {
        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, 0))) {
        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;
 
#ifdef TEST_FRAMEWORK
    /* 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>";
#endif
 
    ast_debug(3, "%s: Destroying SIP session\n", ast_sip_session_get_name(session));
 
    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) {
        struct pjsip_dialog *dlg = session->inv_session->dlg;
 
        /* The INVITE session uses the dialog pool for memory, so we need to
         * decrement its reference first before that of the dialog.
         */
 
#ifdef HAVE_PJSIP_INV_SESSION_REF
        pjsip_inv_dec_ref(session->inv_session);
#endif
        pjsip_dlg_dec_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);
 
    /* When a PJSIP INVITE session is created it is created with a reference
     * count of 1, with that reference being managed by the underlying state
     * of the INVITE session itself. When the INVITE session transitions to
     * a DISCONNECTED state that reference is released. This means we can not
     * rely on that reference to ensure the INVITE session remains for the
     * lifetime of our session. To ensure it does we add our own reference
     * and release it when our own session goes away, ensuring that the INVITE
     * session remains for the lifetime of session.
     */
 
#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");
        return NULL;
    }
#endif
 
    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;
    }
 
    session->authentication_challenge_count = 0;
 
    /* Fire session 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);
    session = inv->mod_data[session_module.id];
 
    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(3, "%s: A reINVITE is being challenged\n", ast_sip_session_get_name(session));
        return PJ_FALSE;
    }
    ast_debug(3, "%s: Initial INVITE is being challenged.\n", ast_sip_session_get_name(session));
 
    if (++session->authentication_challenge_count > MAX_RX_CHALLENGES) {
        ast_debug(3, "%s: Initial INVITE reached maximum number of auth attempts.\n", ast_sip_session_get_name(session));
        return PJ_FALSE;
    }
 
    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, "%s: Delay sending BYE because of outstanding transaction...\n",
                ast_sip_session_get_name(session));
            /* 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, NULL, 1);
            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.
 */
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, "%s: Ending session after being deferred\n", ast_sip_session_get_name(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;
}
 
pjsip_dialog *ast_sip_session_get_dialog(const struct ast_sip_session *session)
{
    pjsip_inv_session *inv_session = session->inv_session;
 
    if (!inv_session) {
        return NULL;
    }
 
    return inv_session->dlg;
}
 
pjsip_inv_state ast_sip_session_get_pjsip_inv_state(const struct ast_sip_session *session)
{
    pjsip_inv_session *inv_session = session->inv_session;
 
    if (!inv_session) {
        return PJSIP_INV_STATE_NULL;
    }
 
    return inv_session->state;
}
 
/*! \brief Fetch just the Caller ID number in order of PAI, RPID, From */
static int fetch_callerid_num(struct ast_sip_session *session, pjsip_rx_data *rdata, char *buf, size_t len)
{
    int res = -1;
    struct ast_party_id id;
 
    ast_party_id_init(&id);
    if (!ast_sip_set_id_from_invite(rdata, &id, &session->endpoint->id.self, session->endpoint->id.trust_inbound)) {
        ast_copy_string(buf, id.number.str, len);
        res = 0;
    }
    ast_party_id_free(&id);
    return res;
}
 
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)
{
    char cid_num[AST_CHANNEL_NAME];
    pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
    struct ast_features_pickup_config *pickup_cfg;
    const char *pickupexten;
 
    if (!ast_sip_is_allowed_uri(ruri)) {
        return SIP_GET_DEST_UNSUPPORTED_URI;
    }
 
    ast_copy_pj_str(session->exten, ast_sip_pjsip_uri_get_username(ruri), sizeof(session->exten));
    if (ast_strlen_zero(session->exten)) {
        /* Some SIP devices send an empty extension for PLAR: this should map to s */
        ast_debug(1, "RURI contains no user portion: defaulting to extension 's'\n");
        ast_copy_string(session->exten, "s", 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(NULL); /* session->channel doesn't exist yet, using NULL */
    if (!pickup_cfg) {
        ast_log(LOG_ERROR, "%s: Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n",
            ast_sip_session_get_name(session));
        pickupexten = "";
    } else {
        pickupexten = ast_strdupa(pickup_cfg->pickupexten);
        ao2_ref(pickup_cfg, -1);
    }
 
    fetch_callerid_num(session, rdata, cid_num, sizeof(cid_num));
 
    /* If there's an overlap_context override specified, use that; otherwise, just use the endpoint's context */
 
    if (!strcmp(session->exten, pickupexten) ||
        ast_exists_extension(NULL, S_OR(session->endpoint->overlap_context, session->endpoint->context), session->exten, 1, S_OR(cid_num, NULL))) {
        /*
         * 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, S_OR(session->endpoint->overlap_context, session->endpoint->context), session->exten, 1, S_OR(cid_num, NULL)))) {
        /* Overlap partial match */
        return SIP_GET_DEST_EXTEN_PARTIAL;
    }
 
    return SIP_GET_DEST_EXTEN_NOT_FOUND;
}
 
/*!
 * \internal
 * \brief Process initial answer for an incoming invite
 *
 * This function should only be called during the setup, and handling of a
 * new incoming invite. Most, if not all of the time, this will be called
 * when an error occurs and we need to respond as such.
 *
 * When a SIP session termination code is given for the answer it's assumed
 * this call then will be the final bit of processing before ending session
 * setup. As such, we've been holding a lock, and a reference on the invite
 * session's dialog. So before returning this function removes that reference,
 * and unlocks the dialog.
 *
 * \param inv_session The session on which to answer
 * \param rdata The original request
 * \param answer_code The answer's numeric code
 * \param terminate_code The termination code if the answer fails
 * \param notify Whether or not to call on_state_changed
 *
 * \retval 0 if invite successfully answered, -1 if an error occurred
 */
static int new_invite_initial_answer(pjsip_inv_session *inv_session, pjsip_rx_data *rdata,
    int answer_code, int terminate_code, pj_bool_t notify)
{
    pjsip_tx_data *tdata = NULL;
    int res = 0;
 
    if (inv_session->state != PJSIP_INV_STATE_DISCONNECTED) {
        if (pjsip_inv_initial_answer(
                inv_session, rdata, answer_code, NULL, NULL, &tdata) != PJ_SUCCESS) {
 
            pjsip_inv_terminate(inv_session, terminate_code ? terminate_code : answer_code, notify);
            res = -1;
        } else {
            pjsip_inv_send_msg(inv_session, tdata);
        }
    }
 
    if (answer_code >= 300) {
        /*
         * A session is ending. The dialog has a reference that needs to be
         * removed and holds a lock that needs to be unlocked before returning.
         */
        pjsip_dlg_dec_lock(inv_session->dlg);
    }
 
    return res;
}
 
/*!
 * \internal
 * \brief Create and initialize a pjsip invite session
 *
 * pjsip_inv_session adds, and maintains a reference to the dialog upon a successful
 * invite session creation until the session is destroyed. However, we'll wait to
 * remove the reference that was added for the dialog when it gets created since we're
 * not ready to unlock the dialog in this function.
 *
 * So, if this function successfully returns that means it returns with its newly
 * created, and associated dialog locked and with two references (i.e. dialog's
 * reference count should be 2).
 *
 * \param rdata The request that is starting the dialog
 * \param endpoint A pointer to the endpoint
 *
 * \return A pjsip invite session object
 * \retval NULL on error
 */
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;
    const pj_str_t STR_100REL = { "100rel", 6};
    unsigned int i;
    pj_status_t dlg_status = PJ_EUNKNOWN;
 
    /*
     * If 100rel is set to "peer_supported" on the endpoint and the peer indicated support for 100rel
     * in the Supported header, send 1xx responses reliably by adding PJSIP_INV_REQUIRE_100REL to pjsip_inv_options flags.
     */
    if (endpoint->rel100 == AST_SIP_100REL_PEER_SUPPORTED && rdata->msg_info.supported != NULL) {
        for (i = 0; i < rdata->msg_info.supported->count; ++i) {
            if (pj_stricmp(&rdata->msg_info.supported->values[i], &STR_100REL) == 0) {
                options |= PJSIP_INV_REQUIRE_100REL;
                break;
            }
        }
    }
 
    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_locked(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;
    }
 
    /*
     * The returned dialog holds a lock and has a reference added. Any paths where the
     * dialog invite session is not returned must unlock the dialog and remove its reference.
     */
 
    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);
        /*
         * The acquired dialog holds a lock, and a reference. Since the dialog is not
         * going to be returned here it must first be unlocked and de-referenced. This
         * must be done prior to calling dialog termination.
         */
        pjsip_dlg_dec_lock(dlg);
        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) {
        /* Dialog's lock and a reference are removed in new_invite_initial_answer */
        new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE);
        /* Remove 2nd reference added at inv_session creation */
        pjsip_dlg_dec_session(inv_session->dlg, &session_module);
        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 &&
        ast_sip_is_media_type_in(&ctype_hdr->media, &pjsip_media_type_multipart_mixed,
            &pjsip_media_type_multipart_alternative, SENTINEL)) {
        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(3, "%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_trace_log(-1, LOG_ERROR, "%s: Session already DISCONNECTED [reason=%d (%s)]\n",
            ast_sip_session_get_name(invite->session),
            invite->session->inv_session->cause,
            pjsip_get_status_text(invite->session->inv_session->cause)->ptr);
        SCOPE_EXIT_RTN_VALUE(-1);
    }
 
    switch (get_destination(invite->session, invite->rdata)) {
    case SIP_GET_DEST_EXTEN_FOUND:
        /* Things worked. Keep going */
        break;
    case SIP_GET_DEST_UNSUPPORTED_URI:
        ast_trace(-1, "%s: Call (%s:%s) to extension '%s' - unsupported uri\n",
            ast_sip_session_get_name(invite->session),
            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, 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_trace(-1, "%s: Call (%s:%s) to extension '%s' - partial match\n",
            ast_sip_session_get_name(invite->session),
            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_trace_log(-1, LOG_NOTICE, "%s: Call (%s:%s) to extension '%s' rejected because extension not found in context '%s'.\n",
            ast_sip_session_get_name(invite->session),
            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) {
        if (tdata) {
            pjsip_inv_send_msg(invite->session->inv_session, tdata);
        } else {
            pjsip_inv_terminate(invite->session->inv_session, 500, PJ_TRUE);
        }
        goto end;
    }
 
    ast_trace(-1, "%s: Call (%s:%s) to extension '%s' sending 100 Trying\n",
        ast_sip_session_get_name(invite->session),
        invite->rdata->tp_info.transport->type_name,
        pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
        invite->session->exten);
    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, 0);
    } else {
        /* We are creating a local SDP which is an offer */
        local = create_local_sdp(invite->session->inv_session, invite->session, NULL, 0);
    }
 
    /* 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:
    SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(invite->session));
}
 
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_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");
    }
 
    /*
     * Upon a successful pre_session_setup the associated dialog is returned locked
     * and with an added reference. Well actually two references. One added when the
     * dialog itself was created, and another added when the pjsip invite session was
     * created and the dialog was added to it.
     *
     * In order to ensure the dialog's, and any of its internal attributes, lifetimes
     * we'll hold the lock and maintain the reference throughout the entire new invite
     * handling process. See ast_sip_create_dialog_uas_locked for more details but,
     * basically we do this to make sure a transport failure does not destroy the dialog
     * and/or transaction out from underneath us between pjsip calls. Alternatively, we
     * could probably release the lock if we needed to, but then we'd have to re-lock and
     * check the dialog and transaction prior to every pjsip call.
     *
     * That means any off nominal/failure paths in this function must remove the associated
     * dialog reference added at dialog creation, and remove the lock. As well the
     * referenced pjsip invite session must be "cleaned up", which should also then
     * remove its reference to the dialog at that time.
     *
     * Nominally we'll unlock the dialog, and release the reference when all new invite
     * process handling has successfully completed.
     */
 
    session = ast_sip_session_alloc(endpoint, NULL, inv_session, rdata);
    if (!session) {
        /* Dialog's lock and reference are removed in new_invite_initial_answer */
        if (!new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE)) {
            /* Terminate the session if it wasn't done in the answer */
            pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
        }
        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);
 
    /*
     * The dialog lock and reference added at dialog creation time must be
     * maintained throughout the new invite process. Since we're pretty much
     * done at this point with things it's safe to go ahead and remove the lock
     * and the reference here. See ast_sip_create_dialog_uas_locked for more info.
     *
     * Note, any future functionality added that does work using the dialog must
     * be done before this.
     */
    pjsip_dlg_dec_lock(inv_session->dlg);
 
    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 pj_bool_t session_on_tx_response(pjsip_tx_data *tdata)
{
    pjsip_dialog *dlg = pjsip_tdata_get_dlg(tdata);
    RAII_VAR(struct ast_sip_session *, session, dlg ? ast_sip_dialog_get_session(dlg) : NULL, ao2_cleanup);
    if (session) {
        handle_outgoing_response(session, tdata);
    }
 
    return PJ_SUCCESS;
}
 
static void resend_reinvite(pj_timer_heap_t *timer, pj_timer_entry *entry)
{
    struct ast_sip_session *session = entry->user_data;
 
    ast_debug(3, "%s: re-INVITE collision timer expired.\n",
        ast_sip_session_get_name(session));
 
    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;
    struct ast_sip_session_media_state *pending_media_state = NULL;
    struct ast_sip_session_media_state *active_media_state = NULL;
    const char *session_name = ast_sip_session_get_name(session);
    int use_pending = 0;
    int use_active = 0;
 
    SCOPE_ENTER(3, "%s\n", session_name);
 
    /*
     * If the two media state topologies are the same this means that the session refresh request
     * did not specify a desired topology, so it does not care. If that is the case we don't even
     * pass one in here resulting in the current topology being used.  It's possible though that
     * either one of the topologies could be NULL so we have to test for that before we check for
     * equality.
     */
 
    /* We only want to clone a media state if its topology is not null */
    use_pending = session->pending_media_state->topology != NULL;
    use_active = session->active_media_state->topology != NULL;
 
    /*
     * If both media states have topologies, we can test for equality.  If they're equal we're not going to
     * clone either states.
     */
    if (use_pending && use_active && ast_stream_topology_equal(session->active_media_state->topology, session->pending_media_state->topology)) {
        use_pending = 0;
        use_active = 0;
    }
 
    if (use_pending) {
        pending_media_state = ast_sip_session_media_state_clone(session->pending_media_state);
        if (!pending_media_state) {
            SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to clone pending media state\n", session_name);
        }