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"

#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);

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";
   }
}

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");
   }

   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));
}

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 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_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);
   if (!new_answer) {
      ast_log(LOG_WARNING, "Could not create a new local SDP answer for channel '%s'\n",
         ast_channel_name(session->channel));
      SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create new SDP\n");
   }

   if (on_sdp_creation) {
      if (on_sdp_creation(session, new_answer)) {
         SCOPE_EXIT_RTN_VALUE(-1, "Callback failed\n");
      }
   }

   pjsip_inv_set_sdp_answer(inv_session, new_answer);

   SCOPE_EXIT_RTN_VALUE(0);
}

void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
   handle_outgoing_response(session, tdata);
   pjsip_inv_send_msg(session->inv_session, tdata);
   return;
}

static pj_bool_t session_on_rx_request(pjsip_rx_data *rdata);
static pj_bool_t session_on_rx_response(pjsip_rx_data *rdata);
static void session_on_tsx_state(pjsip_transaction *tsx, pjsip_event *e);

static pjsip_module session_module = {
   .name = {"Session Module", 14},
   .priority = PJSIP_MOD_PRIORITY_APPLICATION,
   .on_rx_request = session_on_rx_request,
   .on_rx_response = session_on_rx_response,
   .on_tsx_state = session_on_tsx_state,
};

/*! \brief Determine whether the SDP provided requires deferral of negotiating or not
 *
 * \retval 1 re-invite should be deferred and resumed later
 * \retval 0 re-invite should not be deferred
 */
static int sdp_requires_deferral(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
{
   int i;

   if (!session->pending_media_state->topology) {
      session->pending_media_state->topology = ast_stream_topology_alloc();
      if (!session->pending_media_state->topology) {
         return -1;
      }
   }

   for (i = 0; i < sdp->media_count; ++i) {
      /* See if there are registered handlers for this media stream type */
      char media[20];
      struct ast_sip_session_sdp_handler *handler;
      RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
      struct ast_stream *existing_stream = NULL;
      struct ast_stream *stream;
      enum ast_media_type type;
      struct ast_sip_session_media *session_media = NULL;
      enum ast_sip_session_sdp_stream_defer res;
      pjmedia_sdp_media *remote_stream = sdp->media[i];

      /* We need a null-terminated version of the media string */
      ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));

      if (session->active_media_state->topology &&
         (i < ast_stream_topology_get_count(session->active_media_state->topology))) {
         existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, i);
      }

      type = ast_media_type_from_str(media);
      stream = ast_stream_alloc(existing_stream ? ast_stream_get_name(existing_stream) : ast_codec_media_type2str(type), type);
      if (!stream) {
         return -1;
      }

      /* As this is only called on an incoming SDP offer before processing it is not possible
       * for streams and their media sessions to exist.
       */
      if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
         ast_stream_free(stream);
         return -1;
      }

      if (existing_stream) {
         const char *stream_label = ast_stream_get_metadata(existing_stream, "SDP:LABEL");

         if (!ast_strlen_zero(stream_label)) {
            ast_stream_set_metadata(stream, "SDP:LABEL", stream_label);
         }
      }

      session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_media_type_from_str(media), i);
      if (!session_media) {
         return -1;
      }

      /* For backwards compatibility with the core the default audio stream is always sendrecv */
      if (!ast_sip_session_is_pending_stream_default(session, stream) || strcmp(media, "audio")) {
         if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
            /* Stream state reflects our state of a stream, so in the case of
             * sendonly and recvonly we store the opposite since that is what ours
             * is.
             */
            ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
         } else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
            ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
         } else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
            ast_stream_set_state(stream, AST_STREAM_STATE_INACTIVE);
         } else {
            ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
         }
      } else {
         ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
      }

      if (session_media->handler) {
         handler = session_media->handler;
         if (handler->defer_incoming_sdp_stream) {
            res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
               sdp->media[i]);
            switch (res) {
            case AST_SIP_SESSION_SDP_DEFER_NOT_HANDLED:
               break;
            case AST_SIP_SESSION_SDP_DEFER_ERROR:
               return 0;
            case AST_SIP_SESSION_SDP_DEFER_NOT_NEEDED:
               break;
            case AST_SIP_SESSION_SDP_DEFER_NEEDED:
               return 1;
            }
         }
         /* Handled by this handler. Move to the next stream */
         continue;
      }

      handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
      if (!handler_list) {
         ast_debug(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))) {
      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 seesion begin handlers */
   handle_session_begin(session);

   /* Avoid unnecessary ref manipulation to return a session */
   ret_session = session;
   session = NULL;
   return ret_session;
}

/*! \brief struct controlling the suspension of the session's serializer. */
struct ast_sip_session_suspender {
   ast_cond_t cond_suspended;
   ast_cond_t cond_complete;
   int suspended;
   int complete;
};

static void sip_session_suspender_dtor(void *vdoomed)
{
   struct ast_sip_session_suspender *doomed = vdoomed;

   ast_cond_destroy(&doomed->cond_suspended);
   ast_cond_destroy(&doomed->cond_complete);
}

/*!
 * \internal
 * \brief Block the session serializer thread task.
 *
 * \param data Pushed serializer task data for suspension.
 *
 * \retval 0
 */
static int sip_session_suspend_task(void *data)
{
   struct ast_sip_session_suspender *suspender = data;

   ao2_lock(suspender);

   /* Signal that the serializer task is now suspended. */
   suspender->suspended = 1;
   ast_cond_signal(&suspender->cond_suspended);

   /* Wait for the serializer suspension to be completed. */
   while (!suspender->complete) {
      ast_cond_wait(&suspender->cond_complete, ao2_object_get_lockaddr(suspender));
   }

   ao2_unlock(suspender);
   ao2_ref(suspender, -1);

   return 0;
}

void ast_sip_session_suspend(struct ast_sip_session *session)
{
   struct ast_sip_session_suspender *suspender;
   int res;

   ast_assert(session->suspended == NULL);

   if (ast_taskprocessor_is_task(session->serializer)) {
      /* I am the session's serializer thread so I cannot suspend. */
      return;
   }

   if (ast_taskprocessor_is_suspended(session->serializer)) {
      /* The serializer already suspended. */
      return;
   }

   suspender = ao2_alloc(sizeof(*suspender), sip_session_suspender_dtor);
   if (!suspender) {
      /* We will just have to hope that the system does not deadlock */
      return;
   }
   ast_cond_init(&suspender->cond_suspended, NULL);
   ast_cond_init(&suspender->cond_complete, NULL);

   ao2_ref(suspender, +1);
   res = ast_sip_push_task(session->serializer, sip_session_suspend_task, suspender);
   if (res) {
      /* We will just have to hope that the system does not deadlock */
      ao2_ref(suspender, -2);
      return;
   }

   session->suspended = suspender;

   /* Wait for the serializer to get suspended. */
   ao2_lock(suspender);
   while (!suspender->suspended) {
      ast_cond_wait(&suspender->cond_suspended, ao2_object_get_lockaddr(suspender));
   }
   ao2_unlock(suspender);

   ast_taskprocessor_suspend(session->serializer);
}

void ast_sip_session_unsuspend(struct ast_sip_session *session)
{
   struct ast_sip_session_suspender *suspender = session->suspended;

   if (!suspender) {
      /* Nothing to do */
      return;
   }
   session->suspended = NULL;

   /* Signal that the serializer task suspension is now complete. */
   ao2_lock(suspender);
   suspender->complete = 1;
   ast_cond_signal(&suspender->cond_complete);
   ao2_unlock(suspender);

   ao2_ref(suspender, -1);

   ast_taskprocessor_unsuspend(session->serializer);
}

/*!
 * \internal
 * \brief Handle initial INVITE challenge response message.
 * \since 13.5.0
 *
 * \param rdata PJSIP receive response message data.
 *
 * \retval PJ_FALSE Did not handle message.
 * \retval PJ_TRUE Handled message.
 */
static pj_bool_t outbound_invite_auth(pjsip_rx_data *rdata)
{
   pjsip_transaction *tsx;
   pjsip_dialog *dlg;
   pjsip_inv_session *inv;
   pjsip_tx_data *tdata;
   struct ast_sip_session *session;

   if (rdata->msg_info.msg->line.status.code != 401
      && rdata->msg_info.msg->line.status.code != 407) {
      /* Doesn't pertain to us. Move on */
      return PJ_FALSE;
   }

   tsx = pjsip_rdata_get_tsx(rdata);
   dlg = pjsip_rdata_get_dlg(rdata);
   if (!dlg || !tsx) {
      return PJ_FALSE;
   }

   if (tsx->method.id != PJSIP_INVITE_METHOD) {
      /* Not an INVITE that needs authentication */
      return PJ_FALSE;
   }

   inv = pjsip_dlg_get_inv_session(dlg);
   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.
 *
 * \return Nothing
 */
static void sip_session_defer_termination_stop_timer(struct ast_sip_session *session)
{
   if (pj_timer_heap_cancel_if_active(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
      &session->scheduled_termination, session->scheduled_termination.id)) {
      ao2_ref(session, -1);
   }
}

void ast_sip_session_defer_termination_cancel(struct ast_sip_session *session)
{
   if (!session->defer_terminate) {
      /* Already canceled or timer fired. */
      return;
   }

   session->defer_terminate = 0;

   if (session->terminate_while_deferred) {
      /* Complete the termination started by the upper layer. */
      ast_sip_session_terminate(session, 0);
   }

   /* Stop the termination timer if it is still running. */
   sip_session_defer_termination_stop_timer(session);
}

void ast_sip_session_end_if_deferred(struct ast_sip_session *session)
{
   if (!session->defer_end) {
      return;
   }

   session->defer_end = 0;

   if (session->ended_while_deferred) {
      /* Complete the session end started by the remote hangup. */
      ast_debug(3, "%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;
}

enum sip_get_destination_result {
   /*! The extension was successfully found */
   SIP_GET_DEST_EXTEN_FOUND,
   /*! The extension specified in the RURI was not found */
   SIP_GET_DEST_EXTEN_NOT_FOUND,
   /*! The extension specified in the RURI was a partial match */
   SIP_GET_DEST_EXTEN_PARTIAL,
   /*! The RURI is of an unsupported scheme */
   SIP_GET_DEST_UNSUPPORTED_URI,
};

/*!
 * \brief Determine where in the dialplan a call should go
 *
 * This uses the username in the request URI to try to match
 * an extension in the endpoint's configured context in order
 * to route the call.
 *
 * \param session The inbound SIP session
 * \param rdata The SIP INVITE
 */
static enum sip_get_destination_result get_destination(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
   pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
   pjsip_sip_uri *sip_ruri;
   struct ast_features_pickup_config *pickup_cfg;
   const char *pickupexten;

   if (!PJSIP_URI_SCHEME_IS_SIP(ruri) && !PJSIP_URI_SCHEME_IS_SIPS(ruri)) {
      return SIP_GET_DEST_UNSUPPORTED_URI;
   }

   sip_ruri = pjsip_uri_get_uri(ruri);
   ast_copy_pj_str(session->exten, &sip_ruri->user, sizeof(session->exten));

   /*
    * We may want to match in the dialplan without any user
    * options getting in the way.
    */
   AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);

   pickup_cfg = ast_get_chan_features_pickup_config(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);
   }

   if (!strcmp(session->exten, pickupexten) ||
      ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, 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, session->endpoint->context, session->exten, 1, 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 endpoint A pointer to the endpoint
 * \param rdata The request that is starting the dialog
 *
 * \retval 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;
   pj_status_t dlg_status = PJ_EUNKNOWN;

   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 &&
      !pj_stricmp2(&ctype_hdr->media.type, "multipart") &&
      (!pj_stricmp2(&ctype_hdr->media.subtype, "mixed") ||
       !pj_stricmp2(&ctype_hdr->media.subtype, "alternative"))) {
      pjsip_multipart_part *part = pjsip_multipart_get_first_part(body);
      while (part != NULL) {
         if (check_content_disposition_in_multipart(part)) {
            return 1;
         }
         part = pjsip_multipart_get_next_part(body, part);
      }
   }
   return 0;
}

static int new_invite(struct new_invite *invite)
{
   pjsip_tx_data *tdata = NULL;
   pjsip_timer_setting timer;
   pjsip_rdata_sdp_info *sdp_info;
   pjmedia_sdp_session *local = NULL;
   char buffer[AST_SOCKADDR_BUFLEN];
   SCOPE_ENTER(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.
    */
   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);
   if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 100, NULL, NULL, &tdata) != PJ_SUCCESS) {
      pjsip_inv_terminate(invite->session->inv_session, 500, PJ_TRUE);
      goto end;
   }
   ast_sip_session_send_response(invite->session, tdata);

   sdp_info = pjsip_rdata_get_sdp_info(invite->rdata);
   if (sdp_info && (sdp_info->sdp_err == PJ_SUCCESS) && sdp_info->sdp) {
      if (handle_incoming_sdp(invite->session, sdp_info->sdp)) {
         tdata = NULL;
         if (pjsip_inv_end_session(invite->session->inv_session, 488, NULL, &tdata) == PJ_SUCCESS
            && tdata) {
            ast_sip_session_send_response(invite->session, tdata);
         }
         goto end;
      }
      /* We are creating a local SDP which is an answer to their offer */
      local = create_local_sdp(invite->session->inv_session, invite->session, sdp_info->sdp);
   } else {
      /* We are creating a local SDP which is an offer */
      local = create_local_sdp(invite->session->inv_session, invite->session, NULL);
   }

   /* If we were unable to create a local SDP terminate the session early, it won't go anywhere */
   if (!local) {
      tdata = NULL;
      if (pjsip_inv_end_session(invite->session->inv_session, 500, NULL, &tdata) == PJ_SUCCESS
         && tdata) {
         ast_sip_session_send_response(invite->session, tdata);
      }
      goto end;
   }

   pjsip_inv_set_local_sdp(invite->session->inv_session, local);
   pjmedia_sdp_neg_set_prefer_remote_codec_order(invite->session->inv_session->neg, PJ_FALSE);
#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
   if (!invite->session->endpoint->preferred_codec_only) {
      pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
   }
#endif

   handle_incoming_request(invite->session, invite->rdata);

end:
   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 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);
      }
   }

   if (use_active) {
      active_media_state = ast_sip_session_media_state_clone(session->active_media_state);
      if (!active_media_state) {
         ast_sip_session_media_state_free(pending_media_state);
         SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to clone active media state\n", session_name);
      }
   }

   if (delay_request(session, NULL, NULL, on_response, 1, DELAYED_METHOD_INVITE, pending_media_state,
      active_media_state, 1)) {
      ast_sip_session_media_state_free(pending_media_state);
      ast_sip_session_media_state_free(active_media_state);
      SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to add delayed request\n", session_name);
   }

   if (pj_timer_entry_running(&session->rescheduled_reinvite)) {
      /* Timer already running.  Something weird is going on. */
      SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: re-INVITE collision while timer running!!!\n", session_name);
   }

   tv.sec = 0;
   if (inv->role == PJSIP_ROLE_UAC) {
      tv.msec = 2100 + ast_random() % 2000;
   } else {
      tv.msec = ast_random() % 2000;
   }
   pj_timer_entry_init(&session->rescheduled_reinvite, 0, session, resend_reinvite);

   ao2_ref(session, +1);
   if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
      &session->rescheduled_reinvite, &tv) != PJ_SUCCESS) {
      ao2_ref(session, -1);
      SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Couldn't schedule timer\n", session_name);
   }

   SCOPE_EXIT_RTN();
}

static void __print_debug_details(const char *function, pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
{
   int id = session_module.id;
   struct ast_sip_session *session = NULL;

   if (!DEBUG_ATLEAST(5)) {
      /* Debug not spamy enough */
      return;
   }

   ast_log(LOG_DEBUG, "Function %s called on event %s\n",
      function, pjsip_event_str(e->type));
   if (!inv) {
      ast_log(LOG_DEBUG, "Transaction %p does not belong to an inv_session?\n", tsx);
      ast_log(LOG_DEBUG, "The transaction state is %s\n",
         pjsip_tsx_state_str(tsx->state));
      return;
   }
   if (id > -1) {
      session = inv->mod_data[session_module.id];
   }
   if (!session) {
      ast_log(LOG_DEBUG, "inv_session %p has no ast session\n", inv);
   } else {
      ast_log(LOG_DEBUG, "The state change pertains to the endpoint '%s(%s)'\n",
         ast_sorcery_object_get_id(session->endpoint),
         session->channel ? ast_channel_name(session->channel) : "");
   }
   if (inv->invite_tsx) {
      ast_log(LOG_DEBUG, "The inv session still has an invite_tsx (%p)\n",
         inv->invite_tsx);
   } else {
      ast_log(LOG_DEBUG, "The inv session does NOT have an invite_tsx\n");
   }
   if (tsx) {
      ast_log(LOG_DEBUG, "The %s %.*s transaction involved in this state change is %p\n",
         pjsip_role_name(tsx->role),
         (int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
         tsx);
      ast_log(LOG_DEBUG, "The current transaction state is %s\n",
         pjsip_tsx_state_str(tsx->state));
      ast_log(LOG_DEBUG, "The transaction state change event is %s\n",
         pjsip_event_str(e->body.tsx_state.type));
   } else {
      ast_log(LOG_DEBUG, "There is no transaction involved in this state change\n");
   }
   ast_log(LOG_DEBUG, "The current inv state is %s\n", pjsip_inv_state_name(inv->state));
}

#define print_debug_details(inv, tsx, e) __print_debug_details(__PRETTY_FUNCTION__, (inv), (tsx), (e))

static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
   struct ast_sip_session_supplement *supplement;
   struct pjsip_request_line req = rdata->msg_info.msg->line.req;
   SCOPE_ENTER(3, "%s: Method is %.*s\n", ast_sip_session_get_name(session), (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));

   AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
      if (supplement->incoming_request && does_method_match(&req.method.name, supplement->method)) {
         if (supplement->incoming_request(session, rdata)) {
            break;
         }
      }
   }

   SCOPE_EXIT("%s\n", ast_sip_session_get_name(session));
}

static void handle_session_begin(struct ast_sip_session *session)
{
   struct ast_sip_session_supplement *iter;

   AST_LIST_TRAVERSE(&session->supplements, iter, next) {
      if (iter->session_begin) {
         iter->session_begin(session);
      }
   }
}

static void handle_session_destroy(struct ast_sip_session *session)
{
   struct ast_sip_session_supplement *iter;

   AST_LIST_TRAVERSE(&session->supplements, iter, next) {
      if (iter->session_destroy) {
         iter->session_destroy(session);
      }
   }
}

static void handle_session_end(struct ast_sip_session *session)
{
   struct ast_sip_session_supplement *iter;

   /* Session is dead.  Notify the supplements. */
   AST_LIST_TRAVERSE(&session->supplements, iter, next) {
      if (iter->session_end) {
         iter->session_end(session);
      }
   }
}

static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata,
      enum ast_sip_session_response_priority response_priority)
{
   struct ast_sip_session_supplement *supplement;
   struct pjsip_status_line status = rdata->msg_info.msg->line.status;
   SCOPE_ENTER(3, "%s: Response is %d %.*s\n", ast_sip_session_get_name(session),
      status.code, (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));

   AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
      if (!(supplement->response_priority & response_priority)) {
         continue;
      }
      if (supplement->incoming_response && does_method_match(&rdata->msg_info.cseq->method.name, supplement->method)) {
         supplement->incoming_response(session, rdata);
      }
   }

   SCOPE_EXIT("%s\n", ast_sip_session_get_name(session));
}

static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
      enum ast_sip_session_response_priority response_priority)
{
   if (rdata->msg_info.msg->type == PJSIP_REQUEST_MSG) {
      handle_incoming_request(session, rdata);
   } else {
      handle_incoming_response(session, rdata, response_priority);
   }

   return 0;
}

static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
   struct ast_sip_session_supplement *supplement;
   struct pjsip_request_line req = tdata->msg->line.req;
   SCOPE_ENTER(3, "%s: Method is %.*s\n", ast_sip_session_get_name(session),
      (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));

   ast_sip_message_apply_transport(session->endpoint->transport, tdata);

   AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
      if (supplement->outgoing_request && does_method_match(&req.method.name, supplement->method)) {
         supplement->outgoing_request(session, tdata);
      }
   }
   SCOPE_EXIT("%s\n", ast_sip_session_get_name(session));
}

static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
   struct ast_sip_session_supplement *supplement;
   struct pjsip_status_line status = tdata->msg->line.status;
   pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
   SCOPE_ENTER(3, "%s: Method is %.*s, Response is %d %.*s\n", ast_sip_session_get_name(session),
      (int) pj_strlen(&cseq->method.name),
      pj_strbuf(&cseq->method.name), status.code, (int) pj_strlen(&status.reason),
      pj_strbuf(&status.reason));


   if (!cseq) {
      SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Cannot send response due to missing sequence header",
         ast_sip_session_get_name(session));
   }

   ast_sip_message_apply_transport(session->endpoint->transport, tdata);

   AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
      if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
         supplement->outgoing_response(session, tdata);
      }
   }

   SCOPE_EXIT("%s\n", ast_sip_session_get_name(session));
}

static int session_end(void *vsession)
{
   struct ast_sip_session *session = vsession;

   /* Stop the scheduled termination */
   sip_session_defer_termination_stop_timer(session);

   /* Session is dead.  Notify the supplements. */
   handle_session_end(session);

   return 0;
}

/*!
 * \internal
 * \brief Complete ending session activities.
 * \since 13.5.0
 *
 * \param vsession Which session to complete stopping.
 *
 * \retval 0 on success.
 * \retval -1 on error.
 */
static int session_end_completion(void *vsession)
{
   struct ast_sip_session *session = vsession;

   ast_sip_dialog_set_serializer(session->inv_session->dlg, NULL);
   ast_sip_dialog_set_endpoint(session->inv_session->dlg, NULL);

   /* Now we can release the ref that was held by session->inv_session */
   ao2_cleanup(session);
   return 0;
}

static int check_request_status(pjsip_inv_session *inv, pjsip_event *e)
{
   struct ast_sip_session *session = inv->mod_data[session_module.id];
   pjsip_transaction *tsx = e->body.tsx_state.tsx;

   if (tsx->status_code != 503 && tsx->status_code != 408) {
      return 0;
   }

   if (!ast_sip_failover_request(tsx->last_tx)) {
      return 0;
   }

   pjsip_inv_uac_restart(inv, PJ_FALSE);
   /*
    * Bump the ref since it will be on a new transaction and
    * we don't want it to go away along with the old transaction.
    */
   pjsip_tx_data_add_ref(tsx->last_tx);
   ast_sip_session_send_request(session, tsx->last_tx);
   return 1;
}

static void handle_incoming_before_media(pjsip_inv_session *inv,
   struct ast_sip_session *session, pjsip_rx_data *rdata)
{
   pjsip_msg *msg;
   ast_debug(3, "%s: Received %s\n", ast_sip_session_get_name(session), rdata->msg_info.msg->type == PJSIP_REQUEST_MSG ?
         "request" : "response");


   handle_incoming(session, rdata, AST_SIP_SESSION_BEFORE_MEDIA);
   msg = rdata->msg_info.msg;
   if (msg->type == PJSIP_REQUEST_MSG
      && msg->line.req.method.id == PJSIP_ACK_METHOD
      && pjmedia_sdp_neg_get_state(inv->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
      pjsip_tx_data *tdata;

      /*
       * SDP negotiation failed on an incoming call that delayed
       * negotiation and then gave us an invalid SDP answer.  We
       * need to send a BYE to end the call because of the invalid
       * SDP answer.
       */
      ast_debug(1,
         "%s: Ending session due to incomplete SDP negotiation.  %s\n",
         ast_sip_session_get_name(session),
         pjsip_rx_data_get_info(rdata));
      if (pjsip_inv_end_session(inv, 400, NULL, &tdata) == PJ_SUCCESS
         && tdata) {
         ast_sip_session_send_request(session, tdata);
      }
   }
}

static void session_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e)
{
   pjsip_event_id_e type;
   struct ast_sip_session *session = inv->mod_data[session_module.id];
   SCOPE_ENTER(1, "%s Event: %s  Inv State: %s\n", ast_sip_session_get_name(session),
      pjsip_event_str(e->type), pjsip_inv_state_name(inv->state));

   if (ast_shutdown_final()) {
      SCOPE_EXIT_RTN("Shutting down\n");
   }

   if (e) {
      print_debug_details(inv, NULL, e);
      type = e->type;
   } else {
      type = PJSIP_EVENT_UNKNOWN;
   }

   session = inv->mod_data[session_module.id];
   if (!session) {
      SCOPE_EXIT_RTN("No session\n");
   }

   switch(type) {
   case PJSIP_EVENT_TX_MSG:
      break;
   case PJSIP_EVENT_RX_MSG:
      handle_incoming_before_media(inv, session, e->body.rx_msg.rdata);
      break;
   case PJSIP_EVENT_TSX_STATE:
      ast_debug(3, "%s: Source of transaction state change is %s\n", ast_sip_session_get_name(session),
         pjsip_event_str(e->body.tsx_state.type));
      /* Transaction state changes are prompted by some other underlying event. */
      switch(e->body.tsx_state.type) {
      case PJSIP_EVENT_TX_MSG:
         break;
      case PJSIP_EVENT_RX_MSG:
         if (!check_request_status(inv, e)) {
            handle_incoming_before_media(inv, session, e->body.tsx_state.src.rdata);
         }
         break;
      case PJSIP_EVENT_TRANSPORT_ERROR:
      case PJSIP_EVENT_TIMER:
         /*
          * Check the request status on transport error or timeout. A transport
          * error can occur when a TCP socket closes and that can be the result
          * of a 503. Also we may need to failover on a timeout (408).
          */
         check_request_status(inv, e);
         break;
      case PJSIP_EVENT_USER:
      case PJSIP_EVENT_UNKNOWN:
      case PJSIP_EVENT_TSX_STATE:
         /* Inception? */
         break;
      }
      break;
   case PJSIP_EVENT_TRANSPORT_ERROR:
   case PJSIP_EVENT_TIMER:
   case PJSIP_EVENT_UNKNOWN:
   case PJSIP_EVENT_USER:
   default:
      break;
   }

   if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
      if (session->defer_end) {
         ast_debug(3, "%s: Deferring session end\n", ast_sip_session_get_name(session));
         session->ended_while_deferred = 1;
         SCOPE_EXIT_RTN("Deferring\n");
      }

      if (ast_sip_push_task(session->serializer, session_end, session)) {
         /* Do it anyway even though this is not the right thread. */
         session_end(session);
      }
   }

   SCOPE_EXIT_RTN();
}

static void session_inv_on_new_session(pjsip_inv_session *inv, pjsip_event *e)
{
   /* XXX STUB */
}

static int session_end_if_disconnected(int id, pjsip_inv_session *inv)
{
   struct ast_sip_session *session;

   if (inv->state != PJSIP_INV_STATE_DISCONNECTED) {
      return 0;
   }

   /*
    * We are locking because ast_sip_dialog_get_session() needs
    * the dialog locked to get the session by other threads.
    */
   pjsip_dlg_inc_lock(inv->dlg);
   session = inv->mod_data[id];
   inv->mod_data[id] = NULL;
   pjsip_dlg_dec_lock(inv->dlg);

   /*
    * Pass the session ref held by session->inv_session to
    * session_end_completion().
    */
   if (session
      && ast_sip_push_task(session->serializer, session_end_completion, session)) {
      /* Do it anyway even though this is not the right thread. */
      session_end_completion(session);
   }

   return 1;
}

static void session_inv_on_tsx_state_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
{
   ast_sip_session_response_cb cb;
   int id = session_module.id;
   pjsip_tx_data *tdata;
   struct ast_sip_session *session = inv->mod_data[session_module.id];
   SCOPE_ENTER(1, "%s TSX State: %s  Inv State: %s\n", ast_sip_session_get_name(session),
      pjsip_tsx_state_str(tsx->state), pjsip_inv_state_name(inv->state));

   if (ast_shutdown_final()) {
      SCOPE_EXIT_RTN("Shutting down\n");
   }

   session = inv->mod_data[id];

   print_debug_details(inv, tsx, e);
   if (!session) {
      /* The session has ended.  Ignore the transaction change. */
      SCOPE_EXIT_RTN("Session ended\n");
   }

   /*
    * If the session is disconnected really nothing else to do unless currently transacting
    * a BYE. If a BYE then hold off destruction until the transaction timeout occurs. This
    * has to be done for BYEs because sometimes the dialog can be in a disconnected
    * state but the BYE request transaction has not yet completed.
    */
   if (tsx->method.id != PJSIP_BYE_METHOD && session_end_if_disconnected(id, inv)) {
      SCOPE_EXIT_RTN("Disconnected\n");
   }

   switch (e->body.tsx_state.type) {
   case PJSIP_EVENT_TX_MSG:
      /* When we create an outgoing request, we do not have access to the transaction that
       * is created. Instead, We have to place transaction-specific data in the tdata. Here,
       * we transfer the data into the transaction. This way, when we receive a response, we
       * can dig this data out again
       */
      tsx->mod_data[id] = e->body.tsx_state.src.tdata->mod_data[id];
      break;
   case PJSIP_EVENT_RX_MSG:
      cb = ast_sip_mod_data_get(tsx->mod_data, id, MOD_DATA_ON_RESPONSE);
      /* As the PJSIP invite session implementation responds with a 200 OK before we have a
       * chance to be invoked session supplements for BYE requests actually end up executing
       * in the invite session state callback as well. To prevent session supplements from
       * running on the BYE request again we explicitly squash invocation of them here.
       */
      if ((e->body.tsx_state.src.rdata->msg_info.msg->type != PJSIP_REQUEST_MSG) ||
         (tsx->method.id != PJSIP_BYE_METHOD)) {
         handle_incoming(session, e->body.tsx_state.src.rdata,
            AST_SIP_SESSION_AFTER_MEDIA);
      }
      if (tsx->method.id == PJSIP_INVITE_METHOD) {
         if (tsx->role == PJSIP_ROLE_UAC) {
            if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
               /* This means we got a non 2XX final response to our outgoing INVITE */
               if (tsx->status_code == PJSIP_SC_REQUEST_PENDING) {
                  reschedule_reinvite(session, cb);
                  SCOPE_EXIT_RTN("Non 2XX final response\n");
               }
               if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
                  ast_debug(1, "%s: reINVITE received final response code %d\n",
                     ast_sip_session_get_name(session),
                     tsx->status_code);
                  if ((tsx->status_code == 401 || tsx->status_code == 407)
                     && ++session->authentication_challenge_count < MAX_RX_CHALLENGES
                     && !ast_sip_create_request_with_auth(
                        &session->endpoint->outbound_auths,
                        e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
                     /* Send authed reINVITE */
                     ast_sip_session_send_request_with_cb(session, tdata, cb);
                     SCOPE_EXIT_RTN("Sending authed reinvite\n");
                  }
                  /* Per RFC3261 14.1 a response to a re-INVITE should only terminate
                   * the dialog if a 481 or 408 occurs. All other responses should leave
                   * the dialog untouched.
                   */
                  if (tsx->status_code == 481 || tsx->status_code == 408) {
                     if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
                        && tdata) {
                        ast_sip_session_send_request(session, tdata);
                     }
                  }
               }
            } else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
               if (!inv->cancelling
                  && inv->role == PJSIP_ROLE_UAC
                  && inv->state == PJSIP_INV_STATE_CONFIRMED
                  && pjmedia_sdp_neg_was_answer_remote(inv->neg)
                  && pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_DONE
                  && (session->channel && ast_channel_hangupcause(session->channel) == AST_CAUSE_BEARERCAPABILITY_NOTAVAIL)
                  ) {
                  /*
                   * We didn't send a CANCEL but the UAS sent us the 200 OK with an invalid or unacceptable codec SDP.
                   * In this case the SDP negotiation is incomplete and PJPROJECT has already sent the ACK.
                   * So, we send the BYE with 503 status code here. And the actual hangup cause code is already set
                   * to AST_CAUSE_BEARERCAPABILITY_NOTAVAIL by the session_inv_on_media_update(), setting the 503
                   * status code doesn't affect to hangup cause code.
                   */
                  ast_debug(1, "Endpoint '%s(%s)': Ending session due to 200 OK with incomplete SDP negotiation.  %s\n",
                     ast_sorcery_object_get_id(session->endpoint),
                     session->channel ? ast_channel_name(session->channel) : "",
                     pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
                  pjsip_inv_end_session(session->inv_session, 503, NULL, &tdata);
                  SCOPE_EXIT_RTN("Incomplete SDP negotiation\n");
               }

               if (inv->cancelling && tsx->status_code == PJSIP_SC_OK) {
                  int sdp_negotiation_done =
                     pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_DONE;

                  /*
                   * We can get here for the following reasons.
                   *
                   * 1) The race condition detailed in RFC5407 section 3.1.2.
                   * We sent a CANCEL at the same time that the UAS sent us a
                   * 200 OK with a valid SDP for the original INVITE.  As a
                   * result, we have now received a 200 OK for a cancelled
                   * call and the SDP negotiation is complete.  We need to
                   * immediately send a BYE to end the dialog.
                   *
                   * 2) We sent a CANCEL and hit the race condition but the
                   * UAS sent us an invalid SDP with the 200 OK.  In this case
                   * the SDP negotiation is incomplete and PJPROJECT has
                   * already sent the BYE for us because of the invalid SDP.
                   */
                  ast_test_suite_event_notify("PJSIP_SESSION_CANCELED",
                     "Endpoint: %s\r\n"
                     "Channel: %s\r\n"
                     "Message: %s\r\n"
                     "SDP: %s",
                     ast_sorcery_object_get_id(session->endpoint),
                     session->channel ? ast_channel_name(session->channel) : "",
                     pjsip_rx_data_get_info(e->body.tsx_state.src.rdata),
                     sdp_negotiation_done ? "complete" : "incomplete");
                  if (!sdp_negotiation_done) {
                     ast_debug(1, "%s: Incomplete SDP negotiation cancelled session.  %s\n",
                        ast_sip_session_get_name(session),
                        pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
                  } else if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
                     && tdata) {
                     ast_debug(1, "%s: Ending session due to RFC5407 race condition.  %s\n",
                        ast_sip_session_get_name(session),
                        pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
                     ast_sip_session_send_request(session, tdata);
                  }
               }
            }
         }
      } else {
         /* All other methods */
         if (tsx->role == PJSIP_ROLE_UAC) {
            if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
               /* This means we got a final response to our outgoing method */
               ast_debug(1, "%s: %.*s received final response code %d\n",
                  ast_sip_session_get_name(session),
                  (int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
                  tsx->status_code);
               if ((tsx->status_code == 401 || tsx->status_code == 407)
                  && ++session->authentication_challenge_count < MAX_RX_CHALLENGES
                  && !ast_sip_create_request_with_auth(
                     &session->endpoint->outbound_auths,
                     e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
                  /* Send authed version of the method */
                  ast_sip_session_send_request_with_cb(session, tdata, cb);
                  SCOPE_EXIT_RTN("Sending authed %.*s\n",
                     (int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name));
               }
            }
         }
      }
      if (cb) {
         cb(session, e->body.tsx_state.src.rdata);
      }
      break;
   case PJSIP_EVENT_TRANSPORT_ERROR:
   case PJSIP_EVENT_TIMER:
      /*
       * The timer event is run by the pjsip monitor thread and not
       * by the session serializer.
       */
      if (session_end_if_disconnected(id, inv)) {
         SCOPE_EXIT_RTN("Disconnected\n");
      }
      break;
   case PJSIP_EVENT_USER:
   case PJSIP_EVENT_UNKNOWN:
   case PJSIP_EVENT_TSX_STATE:
      /* Inception? */
      break;
   }

   if (AST_LIST_EMPTY(&session->delayed_requests)) {
      /* No delayed request pending, so just return */
      SCOPE_EXIT_RTN("Nothing delayed\n");
   }

   if (tsx->method.id == PJSIP_INVITE_METHOD) {
      if (tsx->state == PJSIP_TSX_STATE_PROCEEDING) {
         ast_debug(3, "%s: INVITE delay check. tsx-state:%s\n",
            ast_sip_session_get_name(session),
            pjsip_tsx_state_str(tsx->state));
         check_delayed_requests(session, invite_proceeding);
      } else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
         /*
          * Terminated INVITE transactions always should result in
          * queuing delayed requests, no matter what event caused
          * the transaction to terminate.
          */
         ast_debug(3, "%s: INVITE delay check. tsx-state:%s\n",
            ast_sip_session_get_name(session),
            pjsip_tsx_state_str(tsx->state));
         check_delayed_requests(session, invite_terminated);
      }
   } else if (tsx->role == PJSIP_ROLE_UAC
      && tsx->state == PJSIP_TSX_STATE_COMPLETED
      && !pj_strcmp2(&tsx->method.name, "UPDATE")) {
      ast_debug(3, "%s: UPDATE delay check. tsx-state:%s\n",
         ast_sip_session_get_name(session),
         pjsip_tsx_state_str(tsx->state));
      check_delayed_requests(session, update_completed);
   }

   SCOPE_EXIT_RTN();
}

static int add_sdp_streams(struct ast_sip_session_media *session_media,
   struct ast_sip_session *session, pjmedia_sdp_session *answer,
   const struct pjmedia_sdp_session *remote,
   struct ast_stream *stream)
{
   struct ast_sip_session_sdp_handler *handler = session_media->handler;
   RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
   int res = 0;
   SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
      ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));

   if (handler) {
      /* if an already assigned handler reports a catastrophic error, fail */
      res = handler->create_outgoing_sdp_stream(session, session_media, answer, remote, stream);
      if (res < 0) {
         SCOPE_EXIT_RTN_VALUE(-1, "Coudn't create sdp stream\n");
      }
      SCOPE_EXIT_RTN_VALUE(0, "Had handler\n");
   }

   handler_list = ao2_find(sdp_handlers, ast_codec_media_type2str(session_media->type), OBJ_KEY);
   if (!handler_list) {
      SCOPE_EXIT_RTN_VALUE(0,