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asterisk/res/res_pjsip_session.c
Kevin Harwell 06dada3f01 codec negotiation: add incoming_call_offer_prefs option 
Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.

This patch does the following:

  Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.

  Adds a new ast_sip_session_caps structure that's set for each session media
object.

  Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.

ASTERISK-28756 #close

Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
2020-03-03 14:51:14 -06:00

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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 "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 *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);
}
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);
ao2_cleanup(session_media->caps);
}
struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_session *session,
struct ast_sip_session_media_state *media_state, enum ast_media_type type, int position)
{
struct ast_sip_session_media *session_media = NULL;
/* It is possible for this media state to already contain a session for the stream. If this
* is the case we simply return it.
*/
if (position < AST_VECTOR_SIZE(&media_state->sessions)) {
return AST_VECTOR_GET(&media_state->sessions, position);
}
/* Determine if we can reuse the session media from the active media state if present */
if (position < AST_VECTOR_SIZE(&session->active_media_state->sessions)) {
session_media = AST_VECTOR_GET(&session->active_media_state->sessions, position);
/* A stream can never exist without an accompanying media session */
if (session_media->type == type) {
ao2_ref(session_media, +1);
} else {
session_media = NULL;
}
}
if (!session_media) {
/* No existing media session we can use so create a new one */
session_media = ao2_alloc_options(sizeof(*session_media), session_media_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!session_media) {
return NULL;
}
session_media->encryption = session->endpoint->media.rtp.encryption;
session_media->remote_ice = session->endpoint->media.rtp.ice_support;
session_media->remote_rtcp_mux = session->endpoint->media.rtcp_mux;
session_media->keepalive_sched_id = -1;
session_media->timeout_sched_id = -1;
session_media->type = type;
session_media->stream_num = position;
if (session->endpoint->media.bundle) {
/* This is a new stream so create a new mid based on media type and position, which makes it unique.
* If this is the result of an offer the mid will just end up getting replaced.
*/
if (ast_asprintf(&session_media->mid, "%s-%d", ast_codec_media_type2str(type), position) < 0) {
ao2_ref(session_media, -1);
return NULL;
}
session_media->bundle_group = 0;
/* Some WebRTC clients can't handle an offer to bundle media streams. Instead they expect them to
* already be bundled. Every client handles this scenario though so if WebRTC is enabled just go
* ahead and treat the streams as having already been bundled.
*/
session_media->bundled = session->endpoint->media.webrtc;
} else {
session_media->bundle_group = -1;
}
session_media->caps = ast_sip_session_caps_alloc();
if (!session_media->caps) {
ao2_ref(session_media, -1);
return NULL;
}
}
if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {
ao2_ref(session_media, -1);
return NULL;
}
/* If this stream will be active in some way and it is the first of this type then consider this the default media session to match */
if (!media_state->default_session[type] && ast_stream_get_state(ast_stream_topology_get_stream(media_state->topology, position)) != AST_STREAM_STATE_REMOVED) {
media_state->default_session[type] = session_media;
}
return session_media;
}
static int is_stream_limitation_reached(enum ast_media_type type, const struct ast_sip_endpoint *endpoint, int *type_streams)
{
switch (type) {
case AST_MEDIA_TYPE_AUDIO:
return !(type_streams[type] < endpoint->media.max_audio_streams);
case AST_MEDIA_TYPE_VIDEO:
return !(type_streams[type] < endpoint->media.max_video_streams);
case AST_MEDIA_TYPE_IMAGE:
/* We don't have an option for image (T.38) streams so cap it to one. */
return (type_streams[type] > 0);
case AST_MEDIA_TYPE_UNKNOWN:
case AST_MEDIA_TYPE_TEXT:
default:
/* We don't want any unknown or "other" streams on our endpoint,
* so always just say we've reached the limit
*/
return 1;
}
}
static int get_mid_bundle_group(const pjmedia_sdp_session *sdp, const char *mid)
{
int bundle_group = 0;
int index;
for (index = 0; index < sdp->attr_count; ++index) {
pjmedia_sdp_attr *attr = sdp->attr[index];
char value[pj_strlen(&attr->value) + 1], *mids = value, *attr_mid;
if (pj_strcmp2(&attr->name, "group") || pj_strncmp2(&attr->value, "BUNDLE", 6)) {
continue;
}
ast_copy_pj_str(value, &attr->value, sizeof(value));
/* Skip the BUNDLE at the front */
mids += 7;
while ((attr_mid = strsep(&mids, " "))) {
if (!strcmp(attr_mid, mid)) {
/* The ordering of attributes determines our internal identification of the bundle group based on number,
* with -1 being not in a bundle group. Since this is only exposed internally for response purposes it's
* actually even fine if things move around.
*/
return bundle_group;
}
}
bundle_group++;
}
return -1;
}
static int set_mid_and_bundle_group(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
const pjmedia_sdp_session *sdp,
const struct pjmedia_sdp_media *stream)
{
pjmedia_sdp_attr *attr;
if (!session->endpoint->media.bundle) {
return 0;
}
/* By default on an incoming negotiation we assume no mid and bundle group is present */
ast_free(session_media->mid);
session_media->mid = NULL;
session_media->bundle_group = -1;
session_media->bundled = 0;
/* Grab the media identifier for the stream */
attr = pjmedia_sdp_media_find_attr2(stream, "mid", NULL);
if (!attr) {
return 0;
}
session_media->mid = ast_calloc(1, attr->value.slen + 1);
if (!session_media->mid) {
return 0;
}
ast_copy_pj_str(session_media->mid, &attr->value, attr->value.slen + 1);
/* Determine what bundle group this is part of */
session_media->bundle_group = get_mid_bundle_group(sdp, session_media->mid);
/* If this is actually part of a bundle group then the other side requested or accepted the bundle request */
session_media->bundled = session_media->bundle_group != -1;
return 0;
}
static void set_remote_mslabel_and_stream_group(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
const pjmedia_sdp_session *sdp,
const struct pjmedia_sdp_media *stream,
struct ast_stream *asterisk_stream)
{
int index;
ast_free(session_media->remote_mslabel);
session_media->remote_mslabel = NULL;
for (index = 0; index < stream->attr_count; ++index) {
pjmedia_sdp_attr *attr = stream->attr[index];
char attr_value[pj_strlen(&attr->value) + 1];
char *ssrc_attribute_name, *ssrc_attribute_value = NULL;
char *msid, *tmp = attr_value;
static const pj_str_t STR_msid = { "msid", 4 };
static const pj_str_t STR_ssrc = { "ssrc", 4 };
if (!pj_strcmp(&attr->name, &STR_msid)) {
ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
msid = strsep(&tmp, " ");
session_media->remote_mslabel = ast_strdup(msid);
break;
} else if (!pj_strcmp(&attr->name, &STR_ssrc)) {
ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
if ((ssrc_attribute_name = strchr(attr_value, ' '))) {
/* This has an actual attribute */
*ssrc_attribute_name++ = '\0';
ssrc_attribute_value = strchr(ssrc_attribute_name, ':');
if (ssrc_attribute_value) {
/* Values are actually optional according to the spec */
*ssrc_attribute_value++ = '\0';
}
if (!strcasecmp(ssrc_attribute_name, "mslabel") && !ast_strlen_zero(ssrc_attribute_value)) {
session_media->remote_mslabel = ast_strdup(ssrc_attribute_value);
break;
}
}
}
}
if (ast_strlen_zero(session_media->remote_mslabel)) {
return;
}
/* Iterate through the existing streams looking for a match and if so then group this with it */
for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
struct ast_sip_session_media *group_session_media;
group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
if (ast_strlen_zero(group_session_media->remote_mslabel) ||
strcmp(group_session_media->remote_mslabel, session_media->remote_mslabel)) {
continue;
}
ast_stream_set_group(asterisk_stream, index);
break;
}
}
static void remove_stream_from_bundle(struct ast_sip_session_media *session_media,
struct ast_stream *stream)
{
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
ast_free(session_media->mid);
session_media->mid = NULL;
session_media->bundle_group = -1;
session_media->bundled = 0;
}
static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
{
int i;
int handled = 0;
int type_streams[AST_MEDIA_TYPE_END] = {0};
if (session->inv_session && session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
ast_log(LOG_ERROR, "Failed to handle incoming SDP. Session has been already disconnected\n");
return -1;
}
/* 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) {
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_sip_session_media *session_media = NULL;
int res;
enum ast_media_type type;
struct ast_stream *stream = NULL;
pjmedia_sdp_media *remote_stream = sdp->media[i];
/* We need a null-terminated version of the media string */
ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));
type = ast_media_type_from_str(media);
/* See if we have an already existing stream, which can occur from SDP deferral checking */
if (i < ast_stream_topology_get_count(session->pending_media_state->topology)) {
stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
}
if (!stream) {
struct ast_stream *existing_stream = NULL;
if (session->active_media_state->topology &&
(i < ast_stream_topology_get_count(session->active_media_state->topology))) {
existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, i);
}
stream = ast_stream_alloc(existing_stream ? ast_stream_get_name(existing_stream) : ast_codec_media_type2str(type), type);
if (!stream) {
return -1;
}
if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
ast_stream_free(stream);
return -1;
}
/* For backwards compatibility with the core default streams are always sendrecv */
if (!ast_sip_session_is_pending_stream_default(session, stream)) {
if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
/* Stream state reflects our state of a stream, so in the case of
* sendonly and recvonly we store the opposite since that is what ours
* is.
*/
ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
ast_stream_set_state(stream, AST_STREAM_STATE_INACTIVE);
} else {
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
}
} else {
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
}
}
session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_media_type_from_str(media), i);
if (!session_media) {
return -1;
}
/* If this stream is already declined mark it as such, or mark it as such if we've reached the limit */
if (!remote_stream->desc.port || is_stream_limitation_reached(type, session->endpoint, type_streams)) {
ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
ast_codec_media_type2str(type), i);
remove_stream_from_bundle(session_media, stream);
continue;
}
set_mid_and_bundle_group(session, session_media, sdp, remote_stream);
set_remote_mslabel_and_stream_group(session, session_media, sdp, remote_stream, stream);
if (session_media->handler) {
handler = session_media->handler;
ast_debug(1, "Negotiating incoming SDP media stream '%s' using %s SDP handler\n",
ast_codec_media_type2str(session_media->type),
session_media->handler->id);
res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
if (res < 0) {
/* Catastrophic failure. Abort! */
return -1;
} else if (res == 0) {
ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
ast_codec_media_type2str(type), i);
remove_stream_from_bundle(session_media, stream);
continue;
} else if (res > 0) {
ast_debug(1, "Media stream '%s' handled by %s\n",
ast_codec_media_type2str(session_media->type),
session_media->handler->id);
/* Handled by this handler. Move to the next stream */
handled = 1;
++type_streams[type];
continue;
}
}
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
if (!handler_list) {
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
continue;
}
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
if (handler == session_media->handler) {
continue;
}
ast_debug(1, "Negotiating incoming SDP media stream '%s' using %s SDP handler\n",
ast_codec_media_type2str(session_media->type),
handler->id);
res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
if (res < 0) {
/* Catastrophic failure. Abort! */
return -1;
} else if (res == 0) {
ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
ast_codec_media_type2str(type), i);
remove_stream_from_bundle(session_media, stream);
continue;
} else if (res > 0) {
ast_debug(1, "Media stream '%s' handled by %s\n",
ast_codec_media_type2str(session_media->type),
handler->id);
/* Handled by this handler. Move to the next stream */
session_media_set_handler(session_media, handler);
handled = 1;
++type_streams[type];
break;
}
}
}
if (!handled) {
return -1;
}
return 0;
}
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;
/* For backwards compatibility we only reflect the stream state correctly on
* the non-default streams. This is because the stream state 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 sort of placed on hold too.
*/
if (!ast_sip_session_is_pending_stream_default(session, asterisk_stream)) {
/* 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);
}
/* We need a null-terminated version of the media string */
ast_copy_pj_str(media, &local->media[index]->desc.media, sizeof(media));
set_mid_and_bundle_group(session, session_media, remote, remote->media[index]);
set_remote_mslabel_and_stream_group(session, session_media, remote, remote->media[index], asterisk_stream);
handler = session_media->handler;
if (handler) {
ast_debug(1, "Applying negotiated SDP media stream '%s' using %s SDP handler\n",
ast_codec_media_type2str(session_media->type),
handler->id);
res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
if (res >= 0) {
ast_debug(1, "Applied negotiated SDP media stream '%s' using %s SDP handler\n",
ast_codec_media_type2str(session_media->type),
handler->id);
return 0;
}
return -1;
}
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
if (!handler_list) {
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
return -1;
}
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
if (handler == session_media->handler) {
continue;
}
ast_debug(1, "Applying negotiated SDP media stream '%s' using %s SDP handler\n",
ast_codec_media_type2str(session_media->type),
handler->id);
res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
if (res < 0) {
/* Catastrophic failure. Abort! */
return -1;
}
if (res > 0) {
ast_debug(1, "Applied negotiated SDP media stream '%s' using %s SDP handler\n",
ast_codec_media_type2str(session_media->type),
handler->id);
/* Handled by this handler. Move to the next stream */
session_media_set_handler(session_media, handler);
return 0;
}
}
if (session_media->handler && session_media->handler->stream_stop) {
ast_debug(1, "Stopping SDP media stream '%s' as it is not currently negotiated\n",
ast_codec_media_type2str(session_media->type));
session_media->handler->stream_stop(session_media);
}
return 0;
}
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 */
if (!session->pending_media_state->topology) {
if (session->active_media_state->topology) {
/*
* This happens when we have negotiated media after receiving a 183,
* and we're now receiving a 200 with a new SDP. In this case, there
* is active_media_state, but the pending_media_state has been reset.
*/
struct ast_sip_session_media_state *active_media_state_clone;
active_media_state_clone =
ast_sip_session_media_state_clone(session->active_media_state);
if (!active_media_state_clone) {
ast_log(LOG_WARNING, "Unable to clone active media state for channel '%s'\n",
session->channel ? ast_channel_name(session->channel) : "unknown");
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, "No pending or active media state for channel '%s'\n",
session->channel ? ast_channel_name(session->channel) : "unknown");
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, "Local SDP for channel '%s' contains %d media streams while we expected it to contain %u\n",
session->channel ? ast_channel_name(session->channel) : "unknown",
ast_stream_topology_get_count(session->pending_media_state->topology), local->media_count);
return -1;
}
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)) {
return -1;
}
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.
*/
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)) {
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);
}
return 0;
}
#define DATASTORE_BUCKETS 53
#define MEDIA_BUCKETS 7
static void session_datastore_destroy(void *obj)
{
struct ast_datastore *datastore = obj;
/* Using the destroy function (if present) destroy the data */
if (datastore->info->destroy != NULL && datastore->data != NULL) {
datastore->info->destroy(datastore->data);
datastore->data = NULL;
}
ast_free((void *) datastore->uid);
datastore->uid = NULL;
}
struct ast_datastore *ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid)
{
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
char uuid_buf[AST_UUID_STR_LEN];
const char *uid_ptr = uid;
if (!info) {
return NULL;
}
datastore = ao2_alloc(sizeof(*datastore), session_datastore_destroy);
if (!datastore) {
return NULL;
}
datastore->info = info;
if (ast_strlen_zero(uid)) {
/* They didn't provide an ID so we'll provide one ourself */
uid_ptr = ast_uuid_generate_str(uuid_buf, sizeof(uuid_buf));
}
datastore->uid = ast_strdup(uid_ptr);
if (!datastore->uid) {
return NULL;
}
ao2_ref(datastore, +1);
return datastore;
}
int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore)
{
ast_assert(datastore != NULL);
ast_assert(datastore->info != NULL);
ast_assert(ast_strlen_zero(datastore->uid) == 0);
if (!ao2_link(session->datastores, datastore)) {
return -1;
}
return 0;
}
struct ast_datastore *ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name)
{
return ao2_find(session->datastores, name, OBJ_KEY);
}
void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name)
{
ao2_callback(session->datastores, OBJ_KEY | OBJ_UNLINK | OBJ_NODATA, NULL, (void *) name);
}
enum delayed_method {
DELAYED_METHOD_INVITE,
DELAYED_METHOD_UPDATE,
DELAYED_METHOD_BYE,
};
/*!
* \internal
* \brief Convert delayed method enum value to a string.
* \since 13.3.0
*
* \param method Delayed method enum value to convert to a string.
*
* \return String value of delayed method.
*/
static const char *delayed_method2str(enum delayed_method method)
{
const char *str = "<unknown>";
switch (method) {
case DELAYED_METHOD_INVITE:
str = "INVITE";
break;
case DELAYED_METHOD_UPDATE:
str = "UPDATE";
break;
case DELAYED_METHOD_BYE:
str = "BYE";
break;
}
return str;
}
/*!
* \brief Structure used for sending delayed requests
*
* Requests are typically delayed because the current transaction
* state of an INVITE. Once the pending INVITE transaction terminates,
* the delayed request will be sent
*/
struct ast_sip_session_delayed_request {
/*! Method of the request */
enum delayed_method method;
/*! Callback to call when the delayed request is created. */
ast_sip_session_request_creation_cb on_request_creation;
/*! Callback to call when the delayed request SDP is created */
ast_sip_session_sdp_creation_cb on_sdp_creation;
/*! Callback to call when the delayed request receives a response */
ast_sip_session_response_cb on_response;
/*! Whether to generate new SDP */
int generate_new_sdp;
/*! Requested media state for the SDP */
struct ast_sip_session_media_state *media_state;
AST_LIST_ENTRY(ast_sip_session_delayed_request) next;
};
static struct ast_sip_session_delayed_request *delayed_request_alloc(
enum delayed_method method,
ast_sip_session_request_creation_cb on_request_creation,
ast_sip_session_sdp_creation_cb on_sdp_creation,
ast_sip_session_response_cb on_response,
int generate_new_sdp,
struct ast_sip_session_media_state *media_state)
{
struct ast_sip_session_delayed_request *delay = ast_calloc(1, sizeof(*delay));
if (!delay) {
return NULL;
}
delay->method = method;
delay->on_request_creation = on_request_creation;
delay->on_sdp_creation = on_sdp_creation;
delay->on_response = on_response;
delay->generate_new_sdp = generate_new_sdp;
delay->media_state = media_state;
return delay;
}
static void delayed_request_free(struct ast_sip_session_delayed_request *delay)
{
ast_sip_session_media_state_free(delay->media_state);
ast_free(delay);
}
/*!
* \internal
* \brief Send a delayed request
*
* \retval -1 failure
* \retval 0 success
* \retval 1 refresh request not sent as no change would occur
*/
static int send_delayed_request(struct ast_sip_session *session, struct ast_sip_session_delayed_request *delay)
{
int res;
ast_debug(3, "Endpoint '%s(%s)' sending delayed %s request.\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
delayed_method2str(delay->method));
switch (delay->method) {
case DELAYED_METHOD_INVITE:
res = sip_session_refresh(session, delay->on_request_creation,
delay->on_sdp_creation, delay->on_response,
AST_SIP_SESSION_REFRESH_METHOD_INVITE, delay->generate_new_sdp, delay->media_state, 1);
/* Ownership of media state transitions to ast_sip_session_refresh */
delay->media_state = NULL;
return res;
case DELAYED_METHOD_UPDATE:
res = sip_session_refresh(session, delay->on_request_creation,
delay->on_sdp_creation, delay->on_response,
AST_SIP_SESSION_REFRESH_METHOD_UPDATE, delay->generate_new_sdp, delay->media_state, 1);
/* Ownership of media state transitions to ast_sip_session_refresh */
delay->media_state = NULL;
return res;
case DELAYED_METHOD_BYE:
ast_sip_session_terminate(session, 0);
return 0;
}
ast_log(LOG_WARNING, "Don't know how to send delayed %s(%d) request.\n",
delayed_method2str(delay->method), delay->method);
return -1;
}
/*!
* \internal
* \brief The current INVITE transaction is in the PROCEEDING state.
* \since 13.3.0
*
* \param vsession Session object.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int invite_proceeding(void *vsession)
{
struct ast_sip_session *session = vsession;
struct ast_sip_session_delayed_request *delay;
int found = 0;
int res = 0;
AST_LIST_TRAVERSE_SAFE_BEGIN(&session->delayed_requests, delay, next) {
switch (delay->method) {
case DELAYED_METHOD_INVITE:
break;
case DELAYED_METHOD_UPDATE:
AST_LIST_REMOVE_CURRENT(next);
res = send_delayed_request(session, delay);
delayed_request_free(delay);
if (!res) {
found = 1;
}
break;
case DELAYED_METHOD_BYE:
/* A BYE is pending so don't bother anymore. */
found = 1;
break;
}
if (found) {
break;
}
}
AST_LIST_TRAVERSE_SAFE_END;
ao2_ref(session, -1);
return res;
}
/*!
* \internal
* \brief The current INVITE transaction is in the TERMINATED state.
* \since 13.3.0
*
* \param vsession Session object.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int invite_terminated(void *vsession)
{
struct ast_sip_session *session = vsession;
struct ast_sip_session_delayed_request *delay;
int found = 0;
int res = 0;
int timer_running;
/* re-INVITE collision timer running? */
timer_running = pj_timer_entry_running(&session->rescheduled_reinvite);
AST_LIST_TRAVERSE_SAFE_BEGIN(&session->delayed_requests, delay, next) {
switch (delay->method) {
case DELAYED_METHOD_INVITE:
if (!timer_running) {
found = 1;
}
break;
case DELAYED_METHOD_UPDATE:
case DELAYED_METHOD_BYE:
found = 1;
break;
}
if (found) {
AST_LIST_REMOVE_CURRENT(next);
res = send_delayed_request(session, delay);
delayed_request_free(delay);
if (!res) {
break;
}
}
}
AST_LIST_TRAVERSE_SAFE_END;
ao2_ref(session, -1);
return res;
}
/*!
* \internal
* \brief INVITE collision timeout.
* \since 13.3.0
*
* \param vsession Session object.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int invite_collision_timeout(void *vsession)
{
struct ast_sip_session *session = vsession;
int res;
if (session->inv_session->invite_tsx) {
/*
* INVITE transaction still active. Let it send
* the collision re-INVITE when it terminates.
*/
ao2_ref(session, -1);
res = 0;
} else {
res = invite_terminated(session);
}
return res;
}
/*!
* \internal
* \brief The current UPDATE transaction is in the COMPLETED state.
* \since 13.3.0
*
* \param vsession Session object.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int update_completed(void *vsession)
{
struct ast_sip_session *session = vsession;
int res;
if (session->inv_session->invite_tsx) {
res = invite_proceeding(session);
} else {
res = invite_terminated(session);
}
return res;
}
static void check_delayed_requests(struct ast_sip_session *session,
int (*cb)(void *vsession))
{
ao2_ref(session, +1);
if (ast_sip_push_task(session->serializer, cb, session)) {
ao2_ref(session, -1);
}
}
static int delay_request(struct ast_sip_session *session,
ast_sip_session_request_creation_cb on_request,
ast_sip_session_sdp_creation_cb on_sdp_creation,
ast_sip_session_response_cb on_response,
int generate_new_sdp,
enum delayed_method method,
struct ast_sip_session_media_state *media_state)
{
struct ast_sip_session_delayed_request *delay = delayed_request_alloc(method,
on_request, on_sdp_creation, on_response, generate_new_sdp, media_state);
if (!delay) {
ast_sip_session_media_state_free(media_state);
return -1;
}
if (method == DELAYED_METHOD_BYE) {
/* Send BYE as early as possible */
AST_LIST_INSERT_HEAD(&session->delayed_requests, delay, next);
} else {
AST_LIST_INSERT_TAIL(&session->delayed_requests, delay, next);
}
return 0;
}
static pjmedia_sdp_session *generate_session_refresh_sdp(struct ast_sip_session *session)
{
pjsip_inv_session *inv_session = session->inv_session;
const pjmedia_sdp_session *previous_sdp = NULL;
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);
}
}
return create_local_sdp(inv_session, session, previous_sdp);
}
static void set_from_header(struct ast_sip_session *session)
{
struct ast_party_id effective_id;
struct ast_party_id connected_id;
pj_pool_t *dlg_pool;
pjsip_fromto_hdr *dlg_info;
pjsip_contact_hdr *dlg_contact;
pjsip_name_addr *dlg_info_name_addr;
pjsip_sip_uri *dlg_info_uri;
pjsip_sip_uri *dlg_contact_uri;
int restricted;
const char *pjsip_from_domain;
if (!session->channel || session->saved_from_hdr) {
return;
}
/* We need to save off connected_id for RPID/PAI generation */
ast_party_id_init(&connected_id);
ast_channel_lock(session->channel);
effective_id = ast_channel_connected_effective_id(session->channel);
ast_party_id_copy(&connected_id, &effective_id);
ast_channel_unlock(session->channel);
restricted =
((ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED);
/* Now set up dlg->local.info so pjsip can correctly generate From */
dlg_pool = session->inv_session->dlg->pool;
dlg_info = session->inv_session->dlg->local.info;
dlg_contact = session->inv_session->dlg->local.contact;
dlg_info_name_addr = (pjsip_name_addr *) dlg_info->uri;
dlg_info_uri = pjsip_uri_get_uri(dlg_info_name_addr);
dlg_contact_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(dlg_contact->uri);
if (session->endpoint->id.trust_outbound || !restricted) {
ast_sip_modify_id_header(dlg_pool, dlg_info, &connected_id);
if (ast_sip_get_use_callerid_contact() && ast_strlen_zero(session->endpoint->contact_user)) {
pj_strdup2(dlg_pool, &dlg_contact_uri->user, S_COR(connected_id.number.valid, connected_id.number.str, ""));
}
}
ast_party_id_free(&connected_id);
if (!ast_strlen_zero(session->endpoint->fromuser)) {
dlg_info_name_addr->display.ptr = NULL;
dlg_info_name_addr->display.slen = 0;
pj_strdup2(dlg_pool, &dlg_info_uri->user, session->endpoint->fromuser);
}
if (!ast_strlen_zero(session->endpoint->fromdomain)) {
pj_strdup2(dlg_pool, &dlg_info_uri->host, session->endpoint->fromdomain);
}
/*
* Channel variable for compatibility with chan_sip SIPFROMDOMAIN
*/
ast_channel_lock(session->channel);
pjsip_from_domain = pbx_builtin_getvar_helper(session->channel, "SIPFROMDOMAIN");
if (!ast_strlen_zero(pjsip_from_domain)) {
ast_debug(3, "From header domain reset by channel variable SIPFROMDOMAIN (%s)\n", pjsip_from_domain);
pj_strdup2(dlg_pool, &dlg_info_uri->host, pjsip_from_domain);
}
ast_channel_unlock(session->channel);
/* We need to save off the non-anonymized From for RPID/PAI generation (for domain) */
session->saved_from_hdr = pjsip_hdr_clone(dlg_pool, dlg_info);
ast_sip_add_usereqphone(session->endpoint, dlg_pool, session->saved_from_hdr->uri);
/* In chan_sip, fromuser and fromdomain trump restricted so we only
* anonymize if they're not set.
*/
if (restricted) {
/* fromuser doesn't provide a display name so we always set it */
pj_strdup2(dlg_pool, &dlg_info_name_addr->display, "Anonymous");
if (ast_strlen_zero(session->endpoint->fromuser)) {
pj_strdup2(dlg_pool, &dlg_info_uri->user, "anonymous");
}
if (ast_sip_get_use_callerid_contact() && ast_strlen_zero(session->endpoint->contact_user)) {
pj_strdup2(dlg_pool, &dlg_contact_uri->user, "anonymous");
}
if (ast_strlen_zero(session->endpoint->fromdomain)) {
pj_strdup2(dlg_pool, &dlg_info_uri->host, "anonymous.invalid");
}
} else {
ast_sip_add_usereqphone(session->endpoint, dlg_pool, dlg_info->uri);
}
}
static int sip_session_refresh(struct ast_sip_session *session,
ast_sip_session_request_creation_cb on_request_creation,
ast_sip_session_sdp_creation_cb on_sdp_creation,
ast_sip_session_response_cb on_response,
enum ast_sip_session_refresh_method method, int generate_new_sdp,
struct ast_sip_session_media_state *media_state,
int queued)
{
pjsip_inv_session *inv_session = session->inv_session;
pjmedia_sdp_session *new_sdp = NULL;
pjsip_tx_data *tdata;
if (media_state && (!media_state->topology || !generate_new_sdp)) {
ast_sip_session_media_state_free(media_state);
return -1;
}
if (inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
/* Don't try to do anything with a hung-up call */
ast_debug(3, "Not sending reinvite to %s because of disconnected state...\n",
ast_sorcery_object_get_id(session->endpoint));
ast_sip_session_media_state_free(media_state);
return 0;
}
/* If the dialog has not yet been established we have to defer until it has */
if (inv_session->dlg->state != PJSIP_DIALOG_STATE_ESTABLISHED) {
ast_debug(3, "Delay sending request to %s because dialog has not been established...\n",
ast_sorcery_object_get_id(session->endpoint));
return delay_request(session, on_request_creation, on_sdp_creation, on_response,
generate_new_sdp,
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE
? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE,
media_state);
}
if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
if (inv_session->invite_tsx) {
/* We can't send a reinvite yet, so delay it */
ast_debug(3, "Delay sending reinvite to %s because of outstanding transaction...\n",
ast_sorcery_object_get_id(session->endpoint));
return delay_request(session, on_request_creation, on_sdp_creation,
on_response, generate_new_sdp, DELAYED_METHOD_INVITE, media_state);
} else if (inv_session->state != PJSIP_INV_STATE_CONFIRMED) {
/* Initial INVITE transaction failed to progress us to a confirmed state
* which means re-invites are not possible
*/
ast_debug(3, "Not sending reinvite to %s because not in confirmed state...\n",
ast_sorcery_object_get_id(session->endpoint));
ast_sip_session_media_state_free(media_state);
return 0;
}
}
if (generate_new_sdp) {
/* SDP can only be generated if current negotiation has already completed */
if (inv_session->neg
&& pjmedia_sdp_neg_get_state(inv_session->neg)
!= PJMEDIA_SDP_NEG_STATE_DONE) {
ast_debug(3, "Delay session refresh with new SDP to %s because SDP negotiation is not yet done...\n",
ast_sorcery_object_get_id(session->endpoint));
return delay_request(session, on_request_creation, on_sdp_creation,
on_response, generate_new_sdp,
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE
? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE, media_state);
}
/* If an explicitly requested media state has been provided use it instead of any pending one */
if (media_state) {
int index;
int type_streams[AST_MEDIA_TYPE_END] = {0};
struct ast_stream *stream;
/* Media state conveys a desired media state, so if there are outstanding
* delayed requests we need to ensure we go into the queue and not jump
* ahead. If we sent this media state now then updates could go out of
* order.
*/
if (!queued && !AST_LIST_EMPTY(&session->delayed_requests)) {
ast_debug(3, "Delay sending reinvite to %s because of outstanding requests...\n",
ast_sorcery_object_get_id(session->endpoint));
return delay_request(session, on_request_creation, on_sdp_creation,
on_response, generate_new_sdp,
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE
? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE, media_state);
}
/* Prune the media state so the number of streams fit within the configured limits - we do it here
* so that the index of the resulting streams in the SDP match. If we simply left the streams out
* of the SDP when producing it we'd be in trouble. We also enforce formats here for media types that
* are configurable on the endpoint.
*/
for (index = 0; index < ast_stream_topology_get_count(media_state->topology); ++index) {
struct ast_stream *existing_stream = NULL;
stream = ast_stream_topology_get_stream(media_state->topology, index);
if (session->active_media_state->topology &&
index < ast_stream_topology_get_count(session->active_media_state->topology)) {
existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
}
if (is_stream_limitation_reached(ast_stream_get_type(stream), session->endpoint, type_streams)) {
if (index < AST_VECTOR_SIZE(&media_state->sessions)) {
struct ast_sip_session_media *session_media = AST_VECTOR_GET(&media_state->sessions, index);
ao2_cleanup(session_media);
AST_VECTOR_REMOVE(&media_state->sessions, index, 1);
}
ast_stream_topology_del_stream(media_state->topology, index);
/* A stream has potentially moved into our spot so we need to jump back so we process it */
index -= 1;
continue;
}
/* No need to do anything with stream if it's media state is removed */
if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
/* If there is no existing stream we can just not have this stream in the topology at all. */
if (!existing_stream) {
ast_stream_topology_del_stream(media_state->topology, index);
index -= 1;
}
continue;
}
/* Enforce the configured allowed codecs on audio and video streams */
if (ast_stream_get_type(stream) == AST_MEDIA_TYPE_AUDIO || ast_stream_get_type(stream) == AST_MEDIA_TYPE_VIDEO) {
struct ast_format_cap *joint_cap;
joint_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!joint_cap) {
ast_sip_session_media_state_free(media_state);
return 0;
}
ast_format_cap_get_compatible(ast_stream_get_formats(stream), session->endpoint->media.codecs, joint_cap);
if (!ast_format_cap_count(joint_cap)) {
ao2_ref(joint_cap, -1);
if (!existing_stream) {
/* If there is no existing stream we can just not have this stream in the topology
* at all.
*/
ast_stream_topology_del_stream(media_state->topology, index);
index -= 1;
continue;
} else if (ast_stream_get_state(stream) != ast_stream_get_state(existing_stream) ||
strcmp(ast_stream_get_name(stream), ast_stream_get_name(existing_stream))) {
/* If the underlying stream is a different type or different name then we have to
* mark it as removed, as it is replacing an existing stream. We do this so order
* is preserved.
*/
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
continue;
} else {
/* However if the stream is otherwise remaining the same we can keep the formats
* that exist on it already which allows media to continue to flow.
*/
joint_cap = ao2_bump(ast_stream_get_formats(existing_stream));
}
}
ast_stream_set_formats(stream, joint_cap);
ao2_cleanup(joint_cap);
}
++type_streams[ast_stream_get_type(stream)];
}
if (session->active_media_state->topology) {
/* SDP is a fun thing. Take for example the fact that streams are never removed. They just become
* declined. To better handle this in the case where something requests a topology change for fewer
* streams than are currently present we fill in the topology to match the current number of streams
* that are active.
*/
for (index = ast_stream_topology_get_count(media_state->topology);
index < ast_stream_topology_get_count(session->active_media_state->topology); ++index) {
struct ast_stream *cloned;
stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
ast_assert(stream != NULL);
cloned = ast_stream_clone(stream, NULL);
if (!cloned) {
ast_sip_session_media_state_free(media_state);
return -1;
}
ast_stream_set_state(cloned, AST_STREAM_STATE_REMOVED);
if (ast_stream_topology_append_stream(media_state->topology, cloned) < 0) {
ast_stream_free(cloned);
ast_sip_session_media_state_free(media_state);
return -1;
}
}
/* If the resulting media state matches the existing active state don't bother doing a session refresh */
if (ast_stream_topology_equal(session->active_media_state->topology, media_state->topology)) {
ast_sip_session_media_state_free(media_state);
/* For external consumers we return 0 to say success, but internally for
* send_delayed_request we return a separate value to indicate that this
* session refresh would be redundant so we didn't send it
*/
return queued ? 1 : 0;
}
}
ast_sip_session_media_state_free(session->pending_media_state);
session->pending_media_state = media_state;
}
new_sdp = generate_session_refresh_sdp(session);
if (!new_sdp) {
ast_log(LOG_ERROR, "Failed to generate session refresh SDP. Not sending session refresh\n");
ast_sip_session_media_state_reset(session->pending_media_state);
return -1;
}
if (on_sdp_creation) {
if (on_sdp_creation(session, new_sdp)) {
ast_sip_session_media_state_reset(session->pending_media_state);
return -1;
}
}
}
if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
if (pjsip_inv_reinvite(inv_session, NULL, new_sdp, &tdata)) {
ast_log(LOG_WARNING, "Failed to create reinvite properly.\n");
if (generate_new_sdp) {
ast_sip_session_media_state_reset(session->pending_media_state);
}
return -1;
}
} else if (pjsip_inv_update(inv_session, NULL, new_sdp, &tdata)) {
ast_log(LOG_WARNING, "Failed to create UPDATE properly.\n");
if (generate_new_sdp) {
ast_sip_session_media_state_reset(session->pending_media_state);
}
return -1;
}
if (on_request_creation) {
if (on_request_creation(session, tdata)) {
if (generate_new_sdp) {
ast_sip_session_media_state_reset(session->pending_media_state);
}
return -1;
}
}
ast_debug(3, "Sending session refresh SDP via %s to %s\n",
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE ? "re-INVITE" : "UPDATE",
ast_sorcery_object_get_id(session->endpoint));
ast_sip_session_send_request_with_cb(session, tdata, on_response);
return 0;
}
int ast_sip_session_refresh(struct ast_sip_session *session,
ast_sip_session_request_creation_cb on_request_creation,
ast_sip_session_sdp_creation_cb on_sdp_creation,
ast_sip_session_response_cb on_response,
enum ast_sip_session_refresh_method method, int generate_new_sdp,
struct ast_sip_session_media_state *media_state)
{
return sip_session_refresh(session, on_request_creation, on_sdp_creation,
on_response, method, generate_new_sdp, media_state, 0);
}
int ast_sip_session_regenerate_answer(struct ast_sip_session *session,
ast_sip_session_sdp_creation_cb on_sdp_creation)
{
pjsip_inv_session *inv_session = session->inv_session;
pjmedia_sdp_session *new_answer = NULL;
const pjmedia_sdp_session *previous_offer = NULL;
/* 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)));
return -1;
}
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));
return -1;
}
if (on_sdp_creation) {
if (on_sdp_creation(session, new_answer)) {
return -1;
}
}
pjsip_inv_set_sdp_answer(inv_session, new_answer);
return 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 pjsip_module session_module = {
.name = {"Session Module", 14},
.priority = PJSIP_MOD_PRIORITY_APPLICATION,
.on_rx_request = session_on_rx_request,
};
/*! \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;
}
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 default streams are always sendrecv */
if (!ast_sip_session_is_pending_stream_default(session, stream)) {
if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
/* Stream state reflects our state of a stream, so in the case of
* sendonly and recvonly we store the opposite since that is what ours
* is.
*/
ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
ast_stream_set_state(stream, AST_STREAM_STATE_INACTIVE);
} else {
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
}
} else {
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
}
if (session_media->handler) {
handler = session_media->handler;
if (handler->defer_incoming_sdp_stream) {
res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
sdp->media[i]);
switch (res) {
case AST_SIP_SESSION_SDP_DEFER_NOT_HANDLED:
break;
case AST_SIP_SESSION_SDP_DEFER_ERROR:
return 0;
case AST_SIP_SESSION_SDP_DEFER_NOT_NEEDED:
break;
case AST_SIP_SESSION_SDP_DEFER_NEEDED:
return 1;
}
}
/* Handled by this handler. Move to the next stream */
continue;
}
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
if (!handler_list) {
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
continue;
}
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
if (handler == session_media->handler) {
continue;
}
if (!handler->defer_incoming_sdp_stream) {
continue;
}
res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
sdp->media[i]);
switch (res) {
case AST_SIP_SESSION_SDP_DEFER_NOT_HANDLED:
continue;
case AST_SIP_SESSION_SDP_DEFER_ERROR:
session_media_set_handler(session_media, handler);
return 0;
case AST_SIP_SESSION_SDP_DEFER_NOT_NEEDED:
/* Handled by this handler. */
session_media_set_handler(session_media, handler);
break;
case AST_SIP_SESSION_SDP_DEFER_NEEDED:
/* Handled by this handler. */
session_media_set_handler(session_media, handler);
return 1;
}
/* Move to the next stream */
break;
}
}
return 0;
}
static pj_bool_t session_reinvite_on_rx_request(pjsip_rx_data *rdata)
{
pjsip_dialog *dlg;
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
pjsip_rdata_sdp_info *sdp_info;
if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD ||
!(dlg = pjsip_ua_find_dialog(&rdata->msg_info.cid->id, &rdata->msg_info.to->tag, &rdata->msg_info.from->tag, PJ_FALSE)) ||
!(session = ast_sip_dialog_get_session(dlg)) ||
!session->channel) {
return PJ_FALSE;
}
if (session->deferred_reinvite) {
pj_str_t key, deferred_key;
pjsip_tx_data *tdata;
/* We use memory from the new request on purpose so the deferred reinvite pool does not grow uncontrollably */
pjsip_tsx_create_key(rdata->tp_info.pool, &key, PJSIP_ROLE_UAS, &rdata->msg_info.cseq->method, rdata);
pjsip_tsx_create_key(rdata->tp_info.pool, &deferred_key, PJSIP_ROLE_UAS, &session->deferred_reinvite->msg_info.cseq->method,
session->deferred_reinvite);
/* If this is a retransmission ignore it */
if (!pj_strcmp(&key, &deferred_key)) {
return PJ_TRUE;
}
/* Otherwise this is a new re-invite, so reject it */
if (pjsip_dlg_create_response(dlg, rdata, 491, NULL, &tdata) == PJ_SUCCESS) {
if (pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL) != PJ_SUCCESS) {
pjsip_tx_data_dec_ref(tdata);
}
}
return PJ_TRUE;
}
if (!(sdp_info = pjsip_rdata_get_sdp_info(rdata)) ||
(sdp_info->sdp_err != PJ_SUCCESS)) {
return PJ_FALSE;
}
if (!sdp_info->sdp) {
const pjmedia_sdp_session *local;
int i;
ast_queue_unhold(session->channel);
pjmedia_sdp_neg_get_active_local(session->inv_session->neg, &local);
if (!local) {
return PJ_FALSE;
}
/*
* Some devices indicate hold with deferred SDP reinvites (i.e. no SDP in the reinvite).
* When hold is initially indicated, we
* - Receive an INVITE with no SDP
* - Send a 200 OK with SDP, indicating sendrecv in the media streams
* - Receive an ACK with SDP, indicating sendonly in the media streams
*
* At this point, the pjmedia negotiator saves the state of the media direction so that
* if we are to send any offers, we'll offer recvonly in the media streams. This is
* problematic if the device is attempting to unhold, though. If the device unholds
* by sending a reinvite with no SDP, then we will respond with a 200 OK with recvonly.
* According to RFC 3264, if an offerer offers recvonly, then the answerer MUST respond
* with sendonly or inactive. The result of this is that the stream is not off hold.
*
* Therefore, in this case, when we receive a reinvite while the stream is on hold, we
* need to be sure to offer sendrecv. This way, the answerer can respond with sendrecv
* in order to get the stream off hold. If this is actually a different purpose reinvite
* (like a session timer refresh), then the answerer can respond to our sendrecv with
* sendonly, keeping the stream on hold.
*/
for (i = 0; i < local->media_count; ++i) {
pjmedia_sdp_media *m = local->media[i];
pjmedia_sdp_attr *recvonly;
pjmedia_sdp_attr *inactive;
recvonly = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "recvonly", NULL);
inactive = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "inactive", NULL);
if (recvonly || inactive) {
pjmedia_sdp_attr *to_remove = recvonly ?: inactive;
pjmedia_sdp_attr *sendrecv;
pjmedia_sdp_attr_remove(&m->attr_count, m->attr, to_remove);
sendrecv = pjmedia_sdp_attr_create(session->inv_session->pool, "sendrecv", NULL);
pjmedia_sdp_media_add_attr(m, sendrecv);
}
}
return PJ_FALSE;
}
if (!sdp_requires_deferral(session, sdp_info->sdp)) {
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;
if (!(offer = create_local_sdp(session->inv_session, session, NULL))) {
pjsip_inv_terminate(session->inv_session, 500, PJ_FALSE);
return -1;
}
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) {
return -1;
}
return 0;
}
static int datastore_hash(const void *obj, int flags)
{
const struct ast_datastore *datastore = obj;
const char *uid = flags & OBJ_KEY ? obj : datastore->uid;
ast_assert(uid != NULL);
return ast_str_hash(uid);
}
static int datastore_cmp(void *obj, void *arg, int flags)
{
const struct ast_datastore *datastore1 = obj;
const struct ast_datastore *datastore2 = arg;
const char *uid2 = flags & OBJ_KEY ? arg : datastore2->uid;
ast_assert(datastore1->uid != NULL);
ast_assert(uid2 != NULL);
return strcmp(datastore1->uid, uid2) ? 0 : CMP_MATCH | CMP_STOP;
}
static void session_destructor(void *obj)
{
struct ast_sip_session *session = obj;
struct ast_sip_session_delayed_request *delay;
/* We dup the endpoint ID in case the endpoint gets freed out from under us */
const char *endpoint_name = session->endpoint ?
ast_strdupa(ast_sorcery_object_get_id(session->endpoint)) : "<none>";
ast_debug(3, "Destroying SIP session with endpoint %s\n", endpoint_name);
ast_test_suite_event_notify("SESSION_DESTROYING",
"Endpoint: %s\r\n"
"AOR: %s\r\n"
"Contact: %s"
, endpoint_name
, session->aor ? ast_sorcery_object_get_id(session->aor) : "<none>"
, session->contact ? ast_sorcery_object_get_id(session->contact) : "<none>"
);
/* fire session destroy handler */
handle_session_destroy(session);
/* remove all registered supplements */
ast_sip_session_remove_supplements(session);
AST_LIST_HEAD_DESTROY(&session->supplements);
/* remove all saved media stats */
AST_VECTOR_RESET(&session->media_stats, ast_free);
AST_VECTOR_FREE(&session->media_stats);
ast_taskprocessor_unreference(session->serializer);
ao2_cleanup(session->datastores);
ast_sip_session_media_state_free(session->active_media_state);
ast_sip_session_media_state_free(session->pending_media_state);
while ((delay = AST_LIST_REMOVE_HEAD(&session->delayed_requests, next))) {
delayed_request_free(delay);
}
ast_party_id_free(&session->id);
ao2_cleanup(session->endpoint);
ao2_cleanup(session->aor);
ao2_cleanup(session->contact);
ao2_cleanup(session->direct_media_cap);
ast_dsp_free(session->dsp);
if (session->inv_session) {
pjsip_dlg_dec_session(session->inv_session->dlg, &session_module);
}
ast_test_suite_event_notify("SESSION_DESTROYED", "Endpoint: %s", endpoint_name);
}
/*! \brief Destructor for SIP channel */
static void sip_channel_destroy(void *obj)
{
struct ast_sip_channel_pvt *channel = obj;
ao2_cleanup(channel->pvt);
ao2_cleanup(channel->session);
}
struct ast_sip_channel_pvt *ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session)
{
struct ast_sip_channel_pvt *channel = ao2_alloc(sizeof(*channel), sip_channel_destroy);
if (!channel) {
return NULL;
}
ao2_ref(pvt, +1);
channel->pvt = pvt;
ao2_ref(session, +1);
channel->session = session;
return channel;
}
struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint,
struct ast_sip_contact *contact, pjsip_inv_session *inv_session, pjsip_rx_data *rdata)
{
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
struct ast_sip_session *ret_session;
int dsp_features = 0;
session = ao2_alloc(sizeof(*session), session_destructor);
if (!session) {
return NULL;
}
AST_LIST_HEAD_INIT(&session->supplements);
AST_LIST_HEAD_INIT_NOLOCK(&session->delayed_requests);
ast_party_id_init(&session->id);
session->direct_media_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!session->direct_media_cap) {
return NULL;
}
session->datastores = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
DATASTORE_BUCKETS, datastore_hash, NULL, datastore_cmp);
if (!session->datastores) {
return NULL;
}
session->active_media_state = ast_sip_session_media_state_alloc();
if (!session->active_media_state) {
return NULL;
}
session->pending_media_state = ast_sip_session_media_state_alloc();
if (!session->pending_media_state) {
return NULL;
}
if (AST_VECTOR_INIT(&session->media_stats, 1) < 0) {
return NULL;
}
if (endpoint->dtmf == AST_SIP_DTMF_INBAND || endpoint->dtmf == AST_SIP_DTMF_AUTO) {
dsp_features |= DSP_FEATURE_DIGIT_DETECT;
}
if (endpoint->faxdetect) {
dsp_features |= DSP_FEATURE_FAX_DETECT;
}
if (dsp_features) {
session->dsp = ast_dsp_new();
if (!session->dsp) {
return NULL;
}
ast_dsp_set_features(session->dsp, dsp_features);
}
session->endpoint = ao2_bump(endpoint);
if (rdata) {
/*
* We must continue using the serializer that the original
* INVITE came in on for the dialog. There may be
* retransmissions already enqueued in the original
* serializer that can result in reentrancy and message
* sequencing problems.
*/
session->serializer = ast_sip_get_distributor_serializer(rdata);
} else {
char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
/* Create name with seq number appended. */
ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip/outsess/%s",
ast_sorcery_object_get_id(endpoint));
session->serializer = ast_sip_create_serializer(tps_name);
}
if (!session->serializer) {
return NULL;
}
ast_sip_dialog_set_serializer(inv_session->dlg, session->serializer);
ast_sip_dialog_set_endpoint(inv_session->dlg, endpoint);
pjsip_dlg_inc_session(inv_session->dlg, &session_module);
inv_session->mod_data[session_module.id] = ao2_bump(session);
session->contact = ao2_bump(contact);
session->inv_session = inv_session;
session->dtmf = endpoint->dtmf;
session->moh_passthrough = endpoint->moh_passthrough;
if (ast_sip_session_add_supplements(session)) {
/* Release the ref held by session->inv_session */
ao2_ref(session, -1);
return NULL;
}
/* Fire seesion begin handlers */
handle_session_begin(session);
/* Avoid unnecessary ref manipulation to return a session */
ret_session = session;
session = NULL;
return ret_session;
}
/*! \brief struct controlling the suspension of the session's serializer. */
struct ast_sip_session_suspender {
ast_cond_t cond_suspended;
ast_cond_t cond_complete;
int suspended;
int complete;
};
static void sip_session_suspender_dtor(void *vdoomed)
{
struct ast_sip_session_suspender *doomed = vdoomed;
ast_cond_destroy(&doomed->cond_suspended);
ast_cond_destroy(&doomed->cond_complete);
}
/*!
* \internal
* \brief Block the session serializer thread task.
*
* \param data Pushed serializer task data for suspension.
*
* \retval 0
*/
static int sip_session_suspend_task(void *data)
{
struct ast_sip_session_suspender *suspender = data;
ao2_lock(suspender);
/* Signal that the serializer task is now suspended. */
suspender->suspended = 1;
ast_cond_signal(&suspender->cond_suspended);
/* Wait for the serializer suspension to be completed. */
while (!suspender->complete) {
ast_cond_wait(&suspender->cond_complete, ao2_object_get_lockaddr(suspender));
}
ao2_unlock(suspender);
ao2_ref(suspender, -1);
return 0;
}
void ast_sip_session_suspend(struct ast_sip_session *session)
{
struct ast_sip_session_suspender *suspender;
int res;
ast_assert(session->suspended == NULL);
if (ast_taskprocessor_is_task(session->serializer)) {
/* I am the session's serializer thread so I cannot suspend. */
return;
}
if (ast_taskprocessor_is_suspended(session->serializer)) {
/* The serializer already suspended. */
return;
}
suspender = ao2_alloc(sizeof(*suspender), sip_session_suspender_dtor);
if (!suspender) {
/* We will just have to hope that the system does not deadlock */
return;
}
ast_cond_init(&suspender->cond_suspended, NULL);
ast_cond_init(&suspender->cond_complete, NULL);
ao2_ref(suspender, +1);
res = ast_sip_push_task(session->serializer, sip_session_suspend_task, suspender);
if (res) {
/* We will just have to hope that the system does not deadlock */
ao2_ref(suspender, -2);
return;
}
session->suspended = suspender;
/* Wait for the serializer to get suspended. */
ao2_lock(suspender);
while (!suspender->suspended) {
ast_cond_wait(&suspender->cond_suspended, ao2_object_get_lockaddr(suspender));
}
ao2_unlock(suspender);
ast_taskprocessor_suspend(session->serializer);
}
void ast_sip_session_unsuspend(struct ast_sip_session *session)
{
struct ast_sip_session_suspender *suspender = session->suspended;
if (!suspender) {
/* Nothing to do */
return;
}
session->suspended = NULL;
/* Signal that the serializer task suspension is now complete. */
ao2_lock(suspender);
suspender->complete = 1;
ast_cond_signal(&suspender->cond_complete);
ao2_unlock(suspender);
ao2_ref(suspender, -1);
ast_taskprocessor_unsuspend(session->serializer);
}
/*!
* \internal
* \brief Handle initial INVITE challenge response message.
* \since 13.5.0
*
* \param rdata PJSIP receive response message data.
*
* \retval PJ_FALSE Did not handle message.
* \retval PJ_TRUE Handled message.
*/
static pj_bool_t outbound_invite_auth(pjsip_rx_data *rdata)
{
pjsip_transaction *tsx;
pjsip_dialog *dlg;
pjsip_inv_session *inv;
pjsip_tx_data *tdata;
struct ast_sip_session *session;
if (rdata->msg_info.msg->line.status.code != 401
&& rdata->msg_info.msg->line.status.code != 407) {
/* Doesn't pertain to us. Move on */
return PJ_FALSE;
}
tsx = pjsip_rdata_get_tsx(rdata);
dlg = pjsip_rdata_get_dlg(rdata);
if (!dlg || !tsx) {
return PJ_FALSE;
}
if (tsx->method.id != PJSIP_INVITE_METHOD) {
/* Not an INVITE that needs authentication */
return PJ_FALSE;
}
inv = pjsip_dlg_get_inv_session(dlg);
if (PJSIP_INV_STATE_CONFIRMED <= inv->state) {
/*
* We cannot handle reINVITE authentication at this
* time because the reINVITE transaction is still in
* progress.
*/
ast_debug(1, "A reINVITE is being challenged.\n");
return PJ_FALSE;
}
ast_debug(1, "Initial INVITE is being challenged.\n");
session = inv->mod_data[session_module.id];
if (ast_sip_create_request_with_auth(&session->endpoint->outbound_auths, rdata,
tsx->last_tx, &tdata)) {
return PJ_FALSE;
}
/*
* Restart the outgoing initial INVITE transaction to deal
* with authentication.
*/
pjsip_inv_uac_restart(inv, PJ_FALSE);
ast_sip_session_send_request(session, tdata);
return PJ_TRUE;
}
static pjsip_module outbound_invite_auth_module = {
.name = {"Outbound INVITE Auth", 20},
.priority = PJSIP_MOD_PRIORITY_DIALOG_USAGE,
.on_rx_response = outbound_invite_auth,
};
/*!
* \internal
* \brief Setup outbound initial INVITE authentication.
* \since 13.5.0
*
* \param dlg PJSIP dialog to attach outbound authentication.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int setup_outbound_invite_auth(pjsip_dialog *dlg)
{
pj_status_t status;
++dlg->sess_count;
status = pjsip_dlg_add_usage(dlg, &outbound_invite_auth_module, NULL);
--dlg->sess_count;
return status != PJ_SUCCESS ? -1 : 0;
}
struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint,
struct ast_sip_contact *contact, const char *location, const char *request_user,
struct ast_stream_topology *req_topology)
{
const char *uri = NULL;
RAII_VAR(struct ast_sip_aor *, found_aor, NULL, ao2_cleanup);
RAII_VAR(struct ast_sip_contact *, found_contact, NULL, ao2_cleanup);
pjsip_timer_setting timer;
pjsip_dialog *dlg;
struct pjsip_inv_session *inv_session;
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
struct ast_sip_session *ret_session;
/* 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));
return NULL;
}
if (!(dlg = ast_sip_create_dialog_uac(endpoint, uri, request_user))) {
return NULL;
}
if (setup_outbound_invite_auth(dlg)) {
pjsip_dlg_terminate(dlg);
return NULL;
}
if (pjsip_inv_create_uac(dlg, NULL, endpoint->extensions.flags, &inv_session) != PJ_SUCCESS) {
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
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);
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_format_cap *req_cap;
struct ast_format_cap *joint_cap;
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;
}
req_cap = ast_stream_get_formats(req_stream);
joint_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!joint_cap) {
continue;
}
ast_format_cap_get_compatible(req_cap, endpoint->media.codecs, joint_cap);
if (!ast_format_cap_count(joint_cap)) {
ao2_ref(joint_cap, -1);
continue;
}
clone_stream = ast_stream_clone(req_stream, NULL);
if (!clone_stream) {
ao2_ref(joint_cap, -1);
continue;
}
if (ast_stream_get_type(req_stream) == AST_MEDIA_TYPE_AUDIO) {
/*
* By appending codecs from the endpoint after compatible ones this
* guarantees that priority is given to those while also allowing
* translation to occur for non-compatible.
*/
ast_format_cap_append_from_cap(joint_cap,
endpoint->media.codecs, AST_MEDIA_TYPE_AUDIO);
}
ast_stream_set_formats(clone_stream, joint_cap);
ao2_ref(joint_cap, -1);
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);
return NULL;
}
}
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);
return NULL;
}
}
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);
return NULL;
}
/* Avoid unnecessary ref manipulation to return a session */
ret_session = session;
session = NULL;
return 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;
if (session->defer_terminate) {
session->terminate_while_deferred = 1;
return;
}
if (!response) {
response = 603;
}
/* The media sessions need to exist for the lifetime of the underlying channel
* to ensure that anything (such as bridge_native_rtp) has access to them as
* appropriate. Since ast_sip_session_terminate is called by chan_pjsip and other
* places when the session is to be terminated we terminate any existing
* media sessions here.
*/
ast_sip_session_media_stats_save(session, session->active_media_state);
SWAP(session->active_media_state, session->pending_media_state);
ast_sip_session_media_state_reset(session->pending_media_state);
switch (session->inv_session->state) {
case PJSIP_INV_STATE_NULL:
if (!session->inv_session->invite_tsx) {
/*
* Normally, it's pjproject's transaction cleanup that ultimately causes the
* final session reference to be released but if both STATE and invite_tsx are NULL,
* we never created a transaction in the first place. In this case, we need to
* do the cleanup ourselves.
*/
/* Transfer the inv_session session reference to the session_end_task */
session->inv_session->mod_data[session_module.id] = NULL;
pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
session_end(session);
/*
* session_end_completion will cleanup the final session reference unless
* ast_sip_session_terminate's caller is holding one.
*/
session_end_completion(session);
} else {
pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
}
break;
case PJSIP_INV_STATE_CONFIRMED:
if (session->inv_session->invite_tsx) {
ast_debug(3, "Delay sending BYE to %s because of outstanding transaction...\n",
ast_sorcery_object_get_id(session->endpoint));
/* If this is delayed the only thing that will happen is a BYE request so we don't
* actually need to store the response code for when it happens.
*/
delay_request(session, NULL, NULL, NULL, 0, DELAYED_METHOD_BYE, NULL);
break;
}
/* Fall through */
default:
status = pjsip_inv_end_session(session->inv_session, response, NULL, &packet);
if (status == PJ_SUCCESS && packet) {
struct ast_sip_session_delayed_request *delay;
/* Flush any delayed requests so they cannot overlap this transaction. */
while ((delay = AST_LIST_REMOVE_HEAD(&session->delayed_requests, next))) {
delayed_request_free(delay);
}
if (packet->msg->type == PJSIP_RESPONSE_MSG) {
ast_sip_session_send_response(session, packet);
} else {
ast_sip_session_send_request(session, packet);
}
}
break;
}
}
static int session_termination_task(void *data)
{
struct ast_sip_session *session = data;
if (session->defer_terminate) {
session->defer_terminate = 0;
if (session->inv_session) {
ast_sip_session_terminate(session, 0);
}
}
ao2_ref(session, -1);
return 0;
}
static void session_termination_cb(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry)
{
struct ast_sip_session *session = entry->user_data;
if (ast_sip_push_task(session->serializer, session_termination_task, session)) {
ao2_cleanup(session);
}
}
int ast_sip_session_defer_termination(struct ast_sip_session *session)
{
pj_time_val delay = { .sec = 60, };
int res;
/* The session should not have an active deferred termination request. */
ast_assert(!session->defer_terminate);
session->defer_terminate = 1;
session->defer_end = 1;
session->ended_while_deferred = 0;
ao2_ref(session, +1);
pj_timer_entry_init(&session->scheduled_termination, 0, session, session_termination_cb);
res = (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
&session->scheduled_termination, &delay) != PJ_SUCCESS) ? -1 : 0;
if (res) {
session->defer_terminate = 0;
ao2_ref(session, -1);
}
return res;
}
/*!
* \internal
* \brief Stop the defer termination timer if it is still running.
* \since 13.5.0
*
* \param session Which session to stop the timer.
*
* \return Nothing
*/
static void sip_session_defer_termination_stop_timer(struct ast_sip_session *session)
{
if (pj_timer_heap_cancel_if_active(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
&session->scheduled_termination, session->scheduled_termination.id)) {
ao2_ref(session, -1);
}
}
void ast_sip_session_defer_termination_cancel(struct ast_sip_session *session)
{
if (!session->defer_terminate) {
/* Already canceled or timer fired. */
return;
}
session->defer_terminate = 0;
if (session->terminate_while_deferred) {
/* Complete the termination started by the upper layer. */
ast_sip_session_terminate(session, 0);
}
/* Stop the termination timer if it is still running. */
sip_session_defer_termination_stop_timer(session);
}
void ast_sip_session_end_if_deferred(struct ast_sip_session *session)
{
if (!session->defer_end) {
return;
}
session->defer_end = 0;
if (session->ended_while_deferred) {
/* Complete the session end started by the remote hangup. */
ast_debug(3, "Ending session (%p) after being deferred\n", session);
session->ended_while_deferred = 0;
session_end(session);
}
}
struct ast_sip_session *ast_sip_dialog_get_session(pjsip_dialog *dlg)
{
pjsip_inv_session *inv_session = pjsip_dlg_get_inv_session(dlg);
struct ast_sip_session *session;
if (!inv_session ||
!(session = inv_session->mod_data[session_module.id])) {
return NULL;
}
ao2_ref(session, +1);
return session;
}
enum sip_get_destination_result {
/*! The extension was successfully found */
SIP_GET_DEST_EXTEN_FOUND,
/*! The extension specified in the RURI was not found */
SIP_GET_DEST_EXTEN_NOT_FOUND,
/*! The extension specified in the RURI was a partial match */
SIP_GET_DEST_EXTEN_PARTIAL,
/*! The RURI is of an unsupported scheme */
SIP_GET_DEST_UNSUPPORTED_URI,
};
/*!
* \brief Determine where in the dialplan a call should go
*
* This uses the username in the request URI to try to match
* an extension in the endpoint's configured context in order
* to route the call.
*
* \param session The inbound SIP session
* \param rdata The SIP INVITE
*/
static enum sip_get_destination_result get_destination(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
pjsip_sip_uri *sip_ruri;
struct ast_features_pickup_config *pickup_cfg;
const char *pickupexten;
if (!PJSIP_URI_SCHEME_IS_SIP(ruri) && !PJSIP_URI_SCHEME_IS_SIPS(ruri)) {
return SIP_GET_DEST_UNSUPPORTED_URI;
}
sip_ruri = pjsip_uri_get_uri(ruri);
ast_copy_pj_str(session->exten, &sip_ruri->user, sizeof(session->exten));
/*
* We may want to match in the dialplan without any user
* options getting in the way.
*/
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);
pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
if (!pickup_cfg) {
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
pickupexten = "";
} else {
pickupexten = ast_strdupa(pickup_cfg->pickupexten);
ao2_ref(pickup_cfg, -1);
}
if (!strcmp(session->exten, pickupexten) ||
ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, NULL)) {
size_t size = pj_strlen(&sip_ruri->host) + 1;
char *domain = ast_alloca(size);
ast_copy_pj_str(domain, &sip_ruri->host, size);
pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
/*
* Save off the INVITE Request-URI in case it is
* needed: CHANNEL(pjsip,request_uri)
*/
session->request_uri = pjsip_uri_clone(session->inv_session->pool, ruri);
return SIP_GET_DEST_EXTEN_FOUND;
}
/*
* Check for partial match via overlap dialling (if enabled)
*/
if (session->endpoint->allow_overlap && (
!strncmp(session->exten, pickupexten, strlen(session->exten)) ||
ast_canmatch_extension(NULL, session->endpoint->context, session->exten, 1, NULL))) {
/* Overlap partial match */
return SIP_GET_DEST_EXTEN_PARTIAL;
}
return SIP_GET_DEST_EXTEN_NOT_FOUND;
}
static pjsip_inv_session *pre_session_setup(pjsip_rx_data *rdata, const struct ast_sip_endpoint *endpoint)
{
pjsip_tx_data *tdata;
pjsip_dialog *dlg;
pjsip_inv_session *inv_session;
unsigned int options = endpoint->extensions.flags;
pj_status_t dlg_status;
if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, ast_sip_get_pjsip_endpoint(), &tdata) != PJ_SUCCESS) {
if (tdata) {
if (pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL) != PJ_SUCCESS) {
pjsip_tx_data_dec_ref(tdata);
}
} else {
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
}
return NULL;
}
dlg = ast_sip_create_dialog_uas(endpoint, rdata, &dlg_status);
if (!dlg) {
if (dlg_status != PJ_EEXISTS) {
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
}
return NULL;
}
if (pjsip_inv_create_uas(dlg, rdata, NULL, options, &inv_session) != PJ_SUCCESS) {
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
pjsip_dlg_terminate(dlg);
return NULL;
}
#if defined(HAVE_PJSIP_REPLACE_MEDIA_STREAM) || defined(PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE)
inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
#endif
if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
}
pjsip_inv_send_msg(inv_session, tdata);
return NULL;
}
return inv_session;
}
struct new_invite {
/*! \brief Session created for the new INVITE */
struct ast_sip_session *session;
/*! \brief INVITE request itself */
pjsip_rx_data *rdata;
};
static int 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];
/* From this point on, any calls to pjsip_inv_terminate have the last argument as PJ_TRUE
* so that we will be notified so we can destroy the session properly
*/
if (invite->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
invite->session->inv_session->cause,
pjsip_get_status_text(invite->session->inv_session->cause)->ptr);
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(invite->session->inv_session);
#endif
return -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:
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 416, NULL, NULL, &tdata) == PJ_SUCCESS) {
ast_sip_session_send_response(invite->session, tdata);
} else {
pjsip_inv_terminate(invite->session->inv_session, 416, PJ_TRUE);
}
goto end;
case SIP_GET_DEST_EXTEN_PARTIAL:
ast_debug(1, "Call from '%s' (%s:%s) to extension '%s' - partial match\n",
ast_sorcery_object_get_id(invite->session->endpoint),
invite->rdata->tp_info.transport->type_name,
pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
invite->session->exten);
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 484, NULL, NULL, &tdata) == PJ_SUCCESS) {
ast_sip_session_send_response(invite->session, tdata);
} else {
pjsip_inv_terminate(invite->session->inv_session, 484, PJ_TRUE);
}
goto end;
case SIP_GET_DEST_EXTEN_NOT_FOUND:
default:
ast_log(LOG_NOTICE, "Call from '%s' (%s:%s) to extension '%s' rejected because extension not found in context '%s'.\n",
ast_sorcery_object_get_id(invite->session->endpoint),
invite->rdata->tp_info.transport->type_name,
pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
invite->session->exten,
invite->session->endpoint->context);
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 404, NULL, NULL, &tdata) == PJ_SUCCESS) {
ast_sip_session_send_response(invite->session, tdata);
} else {
pjsip_inv_terminate(invite->session->inv_session, 404, PJ_TRUE);
}
goto end;
};
pjsip_timer_setting_default(&timer);
timer.min_se = invite->session->endpoint->extensions.timer.min_se;
timer.sess_expires = invite->session->endpoint->extensions.timer.sess_expires;
pjsip_timer_init_session(invite->session->inv_session, &timer);
/*
* At this point, we've verified what we can that won't take awhile,
* so let's go ahead and send a 100 Trying out to stop any
* retransmissions.
*/
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 100, NULL, NULL, &tdata) != PJ_SUCCESS) {
pjsip_inv_terminate(invite->session->inv_session, 500, PJ_TRUE);
goto end;
}
ast_sip_session_send_response(invite->session, tdata);
sdp_info = pjsip_rdata_get_sdp_info(invite->rdata);
if (sdp_info && (sdp_info->sdp_err == PJ_SUCCESS) && sdp_info->sdp) {
if (handle_incoming_sdp(invite->session, sdp_info->sdp)) {
tdata = NULL;
if (pjsip_inv_end_session(invite->session->inv_session, 488, NULL, &tdata) == PJ_SUCCESS
&& tdata) {
ast_sip_session_send_response(invite->session, tdata);
}
goto end;
}
/* We are creating a local SDP which is an answer to their offer */
local = create_local_sdp(invite->session->inv_session, invite->session, sdp_info->sdp);
} else {
/* We are creating a local SDP which is an offer */
local = create_local_sdp(invite->session->inv_session, invite->session, NULL);
}
/* If we were unable to create a local SDP terminate the session early, it won't go anywhere */
if (!local) {
tdata = NULL;
if (pjsip_inv_end_session(invite->session->inv_session, 500, NULL, &tdata) == PJ_SUCCESS
&& tdata) {
ast_sip_session_send_response(invite->session, tdata);
}
goto end;
}
pjsip_inv_set_local_sdp(invite->session->inv_session, local);
pjmedia_sdp_neg_set_prefer_remote_codec_order(invite->session->inv_session->neg, PJ_FALSE);
#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
if (!invite->session->endpoint->preferred_codec_only) {
pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
}
#endif
handle_incoming_request(invite->session, invite->rdata);
end:
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(invite->session->inv_session);
#endif
return 0;
}
static void handle_new_invite_request(pjsip_rx_data *rdata)
{
RAII_VAR(struct ast_sip_endpoint *, endpoint,
ast_pjsip_rdata_get_endpoint(rdata), ao2_cleanup);
pjsip_tx_data *tdata = NULL;
pjsip_inv_session *inv_session = NULL;
struct ast_sip_session *session;
struct new_invite invite;
ast_assert(endpoint != NULL);
inv_session = pre_session_setup(rdata, endpoint);
if (!inv_session) {
/* pre_session_setup() returns a response on failure */
return;
}
#ifdef HAVE_PJSIP_INV_SESSION_REF
if (pjsip_inv_add_ref(inv_session) != PJ_SUCCESS) {
ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
if (inv_session->state != PJSIP_INV_STATE_DISCONNECTED) {
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
pjsip_inv_send_msg(inv_session, tdata);
}
}
return;
}
#endif
session = ast_sip_session_alloc(endpoint, NULL, inv_session, rdata);
if (!session) {
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
pjsip_inv_send_msg(inv_session, tdata);
}
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(inv_session);
#endif
return;
}
/*
* 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);
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;
}
/*!
* \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;
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
pjsip_inv_session *inv_session;
switch (rdata->msg_info.msg->line.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;
}
return handled;
}
static void resend_reinvite(pj_timer_heap_t *timer, pj_timer_entry *entry)
{
struct ast_sip_session *session = entry->user_data;
ast_debug(3, "Endpoint '%s(%s)' re-INVITE collision timer expired.\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "");
if (AST_LIST_EMPTY(&session->delayed_requests)) {
/* No delayed request pending, so just return */
ao2_ref(session, -1);
return;
}
if (ast_sip_push_task(session->serializer, invite_collision_timeout, session)) {
/*
* Uh oh. We now have nothing in the foreseeable future
* to trigger sending the delayed requests.
*/
ao2_ref(session, -1);
}
}
static void reschedule_reinvite(struct ast_sip_session *session, ast_sip_session_response_cb on_response)
{
pjsip_inv_session *inv = session->inv_session;
pj_time_val tv;
ast_debug(3, "Endpoint '%s(%s)' re-INVITE collision.\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "");
if (delay_request(session, NULL, NULL, on_response, 1, DELAYED_METHOD_INVITE, NULL)) {
return;
}
if (pj_timer_entry_running(&session->rescheduled_reinvite)) {
/* Timer already running. Something weird is going on. */
ast_debug(1, "Endpoint '%s(%s)' re-INVITE collision while timer running!!!\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "");
return;
}
tv.sec = 0;
if (inv->role == PJSIP_ROLE_UAC) {
tv.msec = 2100 + ast_random() % 2000;
} else {
tv.msec = ast_random() % 2000;
}
pj_timer_entry_init(&session->rescheduled_reinvite, 0, session, resend_reinvite);
ao2_ref(session, +1);
if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
&session->rescheduled_reinvite, &tv) != PJ_SUCCESS) {
ao2_ref(session, -1);
}
}
static void __print_debug_details(const char *function, pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
{
int id = session_module.id;
struct ast_sip_session *session = NULL;
if (!DEBUG_ATLEAST(5)) {
/* Debug not spamy enough */
return;
}
ast_log(LOG_DEBUG, "Function %s called on event %s\n",
function, pjsip_event_str(e->type));
if (!inv) {
ast_log(LOG_DEBUG, "Transaction %p does not belong to an inv_session?\n", tsx);
ast_log(LOG_DEBUG, "The transaction state is %s\n",
pjsip_tsx_state_str(tsx->state));
return;
}
if (id > -1) {
session = inv->mod_data[session_module.id];
}
if (!session) {
ast_log(LOG_DEBUG, "inv_session %p has no ast session\n", inv);
} else {
ast_log(LOG_DEBUG, "The state change pertains to the endpoint '%s(%s)'\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "");
}
if (inv->invite_tsx) {
ast_log(LOG_DEBUG, "The inv session still has an invite_tsx (%p)\n",
inv->invite_tsx);
} else {
ast_log(LOG_DEBUG, "The inv session does NOT have an invite_tsx\n");
}
if (tsx) {
ast_log(LOG_DEBUG, "The %s %.*s transaction involved in this state change is %p\n",
pjsip_role_name(tsx->role),
(int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
tsx);
ast_log(LOG_DEBUG, "The current transaction state is %s\n",
pjsip_tsx_state_str(tsx->state));
ast_log(LOG_DEBUG, "The transaction state change event is %s\n",
pjsip_event_str(e->body.tsx_state.type));
} else {
ast_log(LOG_DEBUG, "There is no transaction involved in this state change\n");
}
ast_log(LOG_DEBUG, "The current inv state is %s\n", pjsip_inv_state_name(inv->state));
}
#define print_debug_details(inv, tsx, e) __print_debug_details(__PRETTY_FUNCTION__, (inv), (tsx), (e))
static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
struct ast_sip_session_supplement *supplement;
struct pjsip_request_line req = rdata->msg_info.msg->line.req;
ast_debug(3, "Method is %.*s\n", (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
if (supplement->incoming_request && does_method_match(&req.method.name, supplement->method)) {
if (supplement->incoming_request(session, rdata)) {
break;
}
}
}
}
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;
ast_debug(3, "Response is %d %.*s\n", status.code, (int) pj_strlen(&status.reason),
pj_strbuf(&status.reason));
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
if (!(supplement->response_priority & response_priority)) {
continue;
}
if (supplement->incoming_response && does_method_match(&rdata->msg_info.cseq->method.name, supplement->method)) {
supplement->incoming_response(session, rdata);
}
}
}
static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
enum ast_sip_session_response_priority response_priority)
{
ast_debug(3, "Received %s\n", rdata->msg_info.msg->type == PJSIP_REQUEST_MSG ?
"request" : "response");
if (rdata->msg_info.msg->type == PJSIP_REQUEST_MSG) {
handle_incoming_request(session, rdata);
} else {
handle_incoming_response(session, rdata, response_priority);
}
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;
ast_debug(3, "Method is %.*s\n", (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
ast_sip_message_apply_transport(session->endpoint->transport, tdata);
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
if (supplement->outgoing_request && does_method_match(&req.method.name, supplement->method)) {
supplement->outgoing_request(session, tdata);
}
}
}
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);
if (!cseq) {
ast_log(LOG_ERROR, "Cannot send response due to missing sequence header");
return;
}
ast_debug(3, "Method is %.*s, Response is %d %.*s\n", (int) pj_strlen(&cseq->method.name),
pj_strbuf(&cseq->method.name), status.code, (int) pj_strlen(&status.reason),
pj_strbuf(&status.reason));
ast_sip_message_apply_transport(session->endpoint->transport, tdata);
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
supplement->outgoing_response(session, tdata);
}
}
}
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;
handle_incoming(session, rdata, AST_SIP_SESSION_BEFORE_MEDIA);
msg = rdata->msg_info.msg;
if (msg->type == PJSIP_REQUEST_MSG
&& msg->line.req.method.id == PJSIP_ACK_METHOD
&& pjmedia_sdp_neg_get_state(inv->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
pjsip_tx_data *tdata;
/*
* SDP negotiation failed on an incoming call that delayed
* negotiation and then gave us an invalid SDP answer. We
* need to send a BYE to end the call because of the invalid
* SDP answer.
*/
ast_debug(1,
"Endpoint '%s(%s)': Ending session due to incomplete SDP negotiation. %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_rx_data_get_info(rdata));
if (pjsip_inv_end_session(inv, 400, NULL, &tdata) == PJ_SUCCESS
&& tdata) {
ast_sip_session_send_request(session, tdata);
}
}
}
static void session_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e)
{
struct ast_sip_session *session;
pjsip_event_id_e type;
if (ast_shutdown_final()) {
return;
}
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) {
return;
}
switch(type) {
case PJSIP_EVENT_TX_MSG:
break;
case PJSIP_EVENT_RX_MSG:
handle_incoming_before_media(inv, session, e->body.rx_msg.rdata);
break;
case PJSIP_EVENT_TSX_STATE:
ast_debug(3, "Source of transaction state change is %s\n", pjsip_event_str(e->body.tsx_state.type));
/* Transaction state changes are prompted by some other underlying event. */
switch(e->body.tsx_state.type) {
case PJSIP_EVENT_TX_MSG:
break;
case PJSIP_EVENT_RX_MSG:
if (!check_request_status(inv, e)) {
handle_incoming_before_media(inv, session, e->body.tsx_state.src.rdata);
}
break;
case PJSIP_EVENT_TRANSPORT_ERROR:
case PJSIP_EVENT_TIMER:
/*
* Check the request status on transport error or timeout. A transport
* error can occur when a TCP socket closes and that can be the result
* of a 503. Also we may need to failover on a timeout (408).
*/
check_request_status(inv, e);
break;
case PJSIP_EVENT_USER:
case PJSIP_EVENT_UNKNOWN:
case PJSIP_EVENT_TSX_STATE:
/* Inception? */
break;
}
break;
case PJSIP_EVENT_TRANSPORT_ERROR:
case PJSIP_EVENT_TIMER:
case PJSIP_EVENT_UNKNOWN:
case PJSIP_EVENT_USER:
default:
break;
}
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
if (session->defer_end) {
ast_debug(3, "Deferring session (%p) end\n", session);
session->ended_while_deferred = 1;
return;
}
if (ast_sip_push_task(session->serializer, session_end, session)) {
/* Do it anyway even though this is not the right thread. */
session_end(session);
}
}
}
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;
struct ast_sip_session *session;
pjsip_tx_data *tdata;
if (ast_shutdown_final()) {
return;
}
session = inv->mod_data[id];
print_debug_details(inv, tsx, e);
if (!session) {
/* The session has ended. Ignore the transaction change. */
return;
}
/*
* 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)) {
return;
}
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);
return;
}
if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
ast_debug(1, "reINVITE received final response code %d\n",
tsx->status_code);
if ((tsx->status_code == 401 || tsx->status_code == 407)
&& !ast_sip_create_request_with_auth(
&session->endpoint->outbound_auths,
e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
/* Send authed reINVITE */
ast_sip_session_send_request_with_cb(session, tdata, cb);
return;
}
if (tsx->status_code != 488 && tsx->status_code != 500) {
/* Other reinvite failures (except 488 and 500) result in destroying the session. */
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 && 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.
*
* 3) We didn't send a CANCEL but the UAS sent us an invalid
* SDP with the 200 OK. In this case the SDP negotiation is
* incomplete and PJPROJECT has already sent the BYE for us
* because of the invalid SDP.
*/
ast_test_suite_event_notify("PJSIP_SESSION_CANCELED",
"Endpoint: %s\r\n"
"Channel: %s\r\n"
"Message: %s\r\n"
"SDP: %s",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata),
sdp_negotiation_done ? "complete" : "incomplete");
if (!sdp_negotiation_done) {
ast_debug(1, "Endpoint '%s(%s)': Incomplete SDP negotiation cancelled session. %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
} else if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
&& tdata) {
ast_debug(1, "Endpoint '%s(%s)': Ending session due to RFC5407 race condition. %s\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
ast_sip_session_send_request(session, tdata);
}
}
}
}
} else {
/* All other methods */
if (tsx->role == PJSIP_ROLE_UAC) {
if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
/* This means we got a final response to our outgoing method */
ast_debug(1, "%.*s received final response code %d\n",
(int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
tsx->status_code);
if ((tsx->status_code == 401 || tsx->status_code == 407)
&& !ast_sip_create_request_with_auth(
&session->endpoint->outbound_auths,
e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
/* Send authed version of the method */
ast_sip_session_send_request_with_cb(session, tdata, cb);
return;
}
}
}
}
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)) {
return;
}
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 */
return;
}
if (tsx->method.id == PJSIP_INVITE_METHOD) {
if (tsx->state == PJSIP_TSX_STATE_PROCEEDING) {
ast_debug(3, "Endpoint '%s(%s)' INVITE delay check. tsx-state:%s\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_tsx_state_str(tsx->state));
check_delayed_requests(session, invite_proceeding);
} else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
/*
* Terminated INVITE transactions always should result in
* queuing delayed requests, no matter what event caused
* the transaction to terminate.
*/
ast_debug(3, "Endpoint '%s(%s)' INVITE delay check. tsx-state:%s\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_tsx_state_str(tsx->state));
check_delayed_requests(session, invite_terminated);
}
} else if (tsx->role == PJSIP_ROLE_UAC
&& tsx->state == PJSIP_TSX_STATE_COMPLETED
&& !pj_strcmp2(&tsx->method.name, "UPDATE")) {
ast_debug(3, "Endpoint '%s(%s)' UPDATE delay check. tsx-state:%s\n",
ast_sorcery_object_get_id(session->endpoint),
session->channel ? ast_channel_name(session->channel) : "",
pjsip_tsx_state_str(tsx->state));
check_delayed_requests(session, update_completed);
}
}
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;
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) {
return -1;
}
return 0;
}
handler_list = ao2_find(sdp_handlers, ast_codec_media_type2str(session_media->type), OBJ_KEY);
if (!handler_list) {
return 0;
}
/* no handler for this stream type and we have a list to search */
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
if (handler == session_media->handler) {
continue;
}
res = handler->create_outgoing_sdp_stream(session, session_media, answer, remote, stream);
if (res < 0) {
/* catastrophic error */
return -1;
}
if (res > 0) {
/* Handled by this handler. Move to the next stream */
session_media_set_handler(session_media, handler);
return 0;
}
}
/* streams that weren't handled won't be included in generated outbound SDP */
return 0;
}
/*! \brief Bundle group building structure */
struct sip_session_media_bundle_group {
/*! \brief The media identifiers in this bundle group */
char *mids[PJMEDIA_MAX_SDP_MEDIA];
/*! \brief SDP attribute string */
struct ast_str *attr_string;
};
static int add_bundle_groups(struct ast_sip_session *session, pj_pool_t *pool, pjmedia_sdp_session *answer)
{
pj_str_t stmp;
pjmedia_sdp_attr *attr;
struct sip_session_media_bundle_group bundle_groups[PJMEDIA_MAX_SDP_MEDIA];
int index, mid_id;
struct sip_session_media_bundle_group *bundle_group;
if (session->endpoint->media.webrtc) {
attr = pjmedia_sdp_attr_create(pool, "msid-semantic", pj_cstr(&stmp, "WMS *"));
pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
}
if (!session->endpoint->media.bundle) {
return 0;
}
memset(bundle_groups, 0, sizeof(bundle_groups));
/* Build the bundle group layout so we can then add it to the SDP */
for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
struct ast_sip_session_media *session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
/* If this stream is not part of a bundle group we can't add it */
if (session_media->bundle_group == -1) {
continue;
}
bundle_group = &bundle_groups[session_media->bundle_group];
/* If this is the first mid then we need to allocate the attribute string and place BUNDLE in front */
if (!bundle_group->mids[0]) {
bundle_group->mids[0] = session_media->mid;
bundle_group->attr_string = ast_str_create(64);
if (!bundle_group->attr_string) {
continue;
}
ast_str_set(&bundle_group->attr_string, 0, "BUNDLE %s", session_media->mid);
continue;
}
for (mid_id = 1; mid_id < PJMEDIA_MAX_SDP_MEDIA; ++mid_id) {
if (!bundle_group->mids[mid_id]) {
bundle_group->mids[mid_id] = session_media->mid;
ast_str_append(&bundle_group->attr_string, 0, " %s", session_media->mid);
break;
} else if (!strcmp(bundle_group->mids[mid_id], session_media->mid)) {
break;
}
}
}
/* Add all bundle groups that have mids to the SDP */
for (index = 0; index < PJMEDIA_MAX_SDP_MEDIA; ++index) {
bundle_group = &bundle_groups[index];
if (!bundle_group->attr_string) {
continue;
}
attr = pjmedia_sdp_attr_create(pool, "group", pj_cstr(&stmp, ast_str_buffer(bundle_group->attr_string)));
pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
ast_free(bundle_group->attr_string);
}
return 0;
}
static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer)
{
static const pj_str_t STR_IN = { "IN", 2 };
static const pj_str_t STR_IP4 = { "IP4", 3 };
static const pj_str_t STR_IP6 = { "IP6", 3 };
pjmedia_sdp_session *local;
int i;
int stream;
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
ast_log(LOG_ERROR, "Failed to create session SDP. Session has been already disconnected\n");
return NULL;
}
if (!inv->pool_prov || !(local = PJ_POOL_ZALLOC_T(inv->pool_prov, pjmedia_sdp_session))) {
return NULL;
}
if (!offer) {
local->origin.version = local->origin.id = (pj_uint32_t)(ast_random());
} else {
local->origin.version = offer->origin.version + 1;
local->origin.id = offer->origin.id;
}
pj_strdup2(inv->pool_prov, &local->origin.user, session->endpoint->media.sdpowner);
pj_strdup2(inv->pool_prov, &local->name, session->endpoint->media.sdpsession);
if (!session->pending_media_state->topology || !ast_stream_topology_get_count(session->pending_media_state->topology)) {
/* We've encountered a situation where we have been told to create a local SDP but noone has given us any indication
* of what kind of stream topology they would like. We try to not alter the current state of the SDP negotiation
* by using what is currently negotiated. If this is unavailable we fall back to what is configured on the endpoint.
*/
ast_stream_topology_free(session->pending_media_state->topology);
if (session->active_media_state->topology) {
session->pending_media_state->topology = ast_stream_topology_clone(session->active_media_state->topology);
} else {
session->pending_media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
}
if (!session->pending_media_state->topology) {
return NULL;
}
}
for (i = 0; i < ast_stream_topology_get_count(session->pending_media_state->topology); ++i) {
struct ast_sip_session_media *session_media;
struct ast_stream *stream;
unsigned int streams = local->media_count;
/* This code does not enforce any maximum stream count limitations as that is done on either
* the handling of an incoming SDP offer or on the handling of a session refresh.
*/
stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_stream_get_type(stream), i);
if (!session_media) {
return NULL;
}
if (add_sdp_streams(session_media, session, local, offer, stream)) {
return NULL;
}
/* If a stream was actually added then add any additional details */
if (streams != local->media_count) {
pjmedia_sdp_media *media = local->media[streams];
pj_str_t stmp;
pjmedia_sdp_attr *attr;
/* Add the media identifier if present */
if (!ast_strlen_zero(session_media->mid)) {
attr = pjmedia_sdp_attr_create(inv->pool_prov, "mid", pj_cstr(&stmp, session_media->mid));
pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
}
}
/* Ensure that we never exceed the maximum number of streams PJMEDIA will allow. */
if (local->media_count == PJMEDIA_MAX_SDP_MEDIA) {
break;
}
}
/* Add any bundle groups that are present on the media state */
if (add_bundle_groups(session, inv->pool_prov, local)) {
return NULL;
}
/* Use the connection details of an available media if possible for SDP level */
for (stream = 0; stream < local->media_count; stream++) {
if (!local->media[stream]->conn) {
continue;
}
if (local->conn) {
if (!pj_strcmp(&local->conn->net_type, &local->media[stream]->conn->net_type) &&
!pj_strcmp(&local->conn->addr_type, &local->media[stream]->conn->addr_type) &&
!pj_strcmp(&local->conn->addr, &local->media[stream]->conn->addr)) {
local->media[stream]->conn = NULL;
}
continue;
}
/* This stream's connection info will serve as the connection details for SDP level */
local->conn = local->media[stream]->conn;
local->media[stream]->conn = NULL;
continue;
}
/* If no SDP level connection details are present then create some */
if (!local->conn) {
local->conn = pj_pool_zalloc(inv->pool_prov, sizeof(struct pjmedia_sdp_conn));
local->conn->net_type = STR_IN;
local->conn->addr_type = session->endpoint->media.rtp.ipv6 ? STR_IP6 : STR_IP4;
if (!ast_strlen_zero(session->endpoint->media.address)) {
pj_strdup2(inv->pool_prov, &local->conn->addr, session->endpoint->media.address);
} else {
pj_strdup2(inv->pool_prov, &local->conn->addr, ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET()));
}
}
pj_strassign(&local->origin.net_type, &local->conn->net_type);
pj_strassign(&local->origin.addr_type, &local->conn->addr_type);
pj_strassign(&local->origin.addr, &local->conn->addr);
return local;
}
static void session_inv_on_rx_offer(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
{
struct ast_sip_session *session;
pjmedia_sdp_session *answer;
if (ast_shutdown_final()) {
return;
}
session = inv->mod_data[session_module.id];
if (handle_incoming_sdp(session, offer)) {
return;
}
if ((answer = create_local_sdp(inv, session, offer))) {
pjsip_inv_set_sdp_answer(inv, answer);
}
}
#if 0
static void session_inv_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
/* XXX STUB */
}
#endif
static void session_inv_on_media_update(pjsip_inv_session *inv, pj_status_t status)
{
struct ast_sip_session *session;
const pjmedia_sdp_session *local, *remote;
if (ast_shutdown_final()) {
return;
}
session = inv->mod_data[session_module.id];
if (!session || !session->channel) {
/*
* If we don't have a session or channel then we really
* don't care about media updates.
* Just ignore
*/
return;
}
if (session->endpoint) {
int bail = 0;
/*
* If following_fork is set, then this is probably the result of a
* forked INVITE and SDP asnwers coming from the different fork UAS
* destinations. In this case updated_sdp_answer will also be set.
*
* If only updated_sdp_answer is set, then this is the non-forking
* scenario where the same UAS just needs to change something like
* the media port.
*/
if (inv->following_fork) {
if (session->endpoint->media.rtp.follow_early_media_fork) {
ast_debug(3, "Following early media fork with different To tags\n");
} else {
ast_debug(3, "Not following early media fork with different To tags\n");
bail = 1;
}
}
#ifdef HAVE_PJSIP_INV_ACCEPT_MULTIPLE_SDP_ANSWERS
else if (inv->updated_sdp_answer) {
if (session->endpoint->media.rtp.accept_multiple_sdp_answers) {
ast_debug(3, "Accepting updated SDP with same To tag\n");
} else {
ast_debug(3, "Ignoring updated SDP answer with same To tag\n");
bail = 1;
}
}
#endif
if (bail) {
return;
}
}
if ((status != PJ_SUCCESS) || (pjmedia_sdp_neg_get_active_local(inv->neg, &local) != PJ_SUCCESS) ||
(pjmedia_sdp_neg_get_active_remote(inv->neg, &remote) != PJ_SUCCESS)) {
ast_channel_hangupcause_set(session->channel, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
ast_queue_hangup(session->channel);
return;
}
handle_negotiated_sdp(session, local, remote);
}
static pjsip_redirect_op session_inv_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e)
{
struct ast_sip_session *session;
const pjsip_sip_uri *uri;
if (ast_shutdown_final()) {
return PJSIP_REDIRECT_STOP;
}
session = inv->mod_data[session_module.id];
if (!session || !session->channel) {
return PJSIP_REDIRECT_STOP;
}
if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_PJSIP) {
return PJSIP_REDIRECT_ACCEPT;
}
if (!PJSIP_URI_SCHEME_IS_SIP(target) && !PJSIP_URI_SCHEME_IS_SIPS(target)) {
return PJSIP_REDIRECT_STOP;
}
handle_incoming(session, e->body.rx_msg.rdata, AST_SIP_SESSION_BEFORE_REDIRECTING);
uri = pjsip_uri_get_uri(target);
if (session->endpoint->redirect_method == AST_SIP_REDIRECT_USER) {
char exten[AST_MAX_EXTENSION];
ast_copy_pj_str(exten, &uri->user, sizeof(exten));
/*
* We may want to match in the dialplan without any user
* options getting in the way.
*/
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
ast_channel_call_forward_set(session->channel, exten);
} else if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_CORE) {
char target_uri[PJSIP_MAX_URL_SIZE];
/* PJSIP/ + endpoint length + / + max URL size */
char forward[8 + strlen(ast_sorcery_object_get_id(session->endpoint)) + PJSIP_MAX_URL_SIZE];
pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, target_uri, sizeof(target_uri));
sprintf(forward, "PJSIP/%s/%s", ast_sorcery_object_get_id(session->endpoint), target_uri);
ast_channel_call_forward_set(session->channel, forward);
}
return PJSIP_REDIRECT_STOP;
}
static pjsip_inv_callback inv_callback = {
.on_state_changed = session_inv_on_state_changed,
.on_new_session = session_inv_on_new_session,
.on_tsx_state_changed = session_inv_on_tsx_state_changed,
.on_rx_offer = session_inv_on_rx_offer,
.on_media_update = session_inv_on_media_update,
.on_redirected = session_inv_on_redirected,
};
/*! \brief Hook for modifying outgoing messages with SDP to contain the proper address information */
static void session_outgoing_nat_hook(pjsip_tx_data *tdata, struct ast_sip_transport *transport)
{
RAII_VAR(struct ast_sip_transport_state *, transport_state, ast_sip_get_transport_state(ast_sorcery_object_get_id(transport)), ao2_cleanup);
struct ast_sip_nat_hook *hook = ast_sip_mod_data_get(
tdata->mod_data, session_module.id, MOD_DATA_NAT_HOOK);
struct pjmedia_sdp_session *sdp;
int stream;
/* SDP produced by us directly will never be multipart */
if (!transport_state || hook || !tdata->msg->body ||
!ast_sip_is_content_type(&tdata->msg->body->content_type, "application", "sdp") ||
ast_strlen_zero(transport->external_media_address)) {
return;
}
sdp = tdata->msg->body->data;
if (sdp->conn) {
char host[NI_MAXHOST];
struct ast_sockaddr our_sdp_addr = { { 0, } };
ast_copy_pj_str(host, &sdp->conn->addr, sizeof(host));
ast_sockaddr_parse(&our_sdp_addr, host, PARSE_PORT_FORBID);
/* Reversed check here. We don't check the remote
* endpoint being in our local net, but whether our
* outgoing session IP is local. If it is, we'll do
* rewriting. No localnet configured? Always rewrite. */
if (ast_sip_transport_is_local(transport_state, &our_sdp_addr) || !transport_state->localnet) {
ast_debug(5, "Setting external media address to %s\n", ast_sockaddr_stringify_host(&transport_state->external_media_address));
pj_strdup2(tdata->pool, &sdp->conn->addr, ast_sockaddr_stringify_host(&transport_state->external_media_address));
pj_strassign(&sdp->origin.addr, &sdp->conn->addr);
}
}
for (stream = 0; stream < sdp->media_count; ++stream) {
/* See if there are registered handlers for this media stream type */
char media[20];
struct ast_sip_session_sdp_handler *handler;
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
/* We need a null-terminated version of the media string */
ast_copy_pj_str(media, &sdp->media[stream]->desc.media, sizeof(media));
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
if (!handler_list) {
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
continue;
}
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
if (handler->change_outgoing_sdp_stream_media_address) {
handler->change_outgoing_sdp_stream_media_address(tdata, sdp->media[stream], transport);
}
}
}
/* We purposely do this so that the hook will not be invoked multiple times, ie: if a retransmit occurs */
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, session_module.id, MOD_DATA_NAT_HOOK, nat_hook);
}
static int load_module(void)
{
pjsip_endpoint *endpt;
if (!ast_sip_get_sorcery() || !ast_sip_get_pjsip_endpoint()) {
return AST_MODULE_LOAD_DECLINE;
}
if (!(nat_hook = ast_sorcery_alloc(ast_sip_get_sorcery(), "nat_hook", NULL))) {
return AST_MODULE_LOAD_DECLINE;
}
nat_hook->outgoing_external_message = session_outgoing_nat_hook;
ast_sorcery_create(ast_sip_get_sorcery(), nat_hook);
sdp_handlers = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
SDP_HANDLER_BUCKETS, sdp_handler_list_hash, NULL, sdp_handler_list_cmp);
if (!sdp_handlers) {
return AST_MODULE_LOAD_DECLINE;
}
endpt = ast_sip_get_pjsip_endpoint();
pjsip_inv_usage_init(endpt, &inv_callback);
pjsip_100rel_init_module(endpt);
pjsip_timer_init_module(endpt);
if (ast_sip_register_service(&session_module)) {
return AST_MODULE_LOAD_DECLINE;
}
ast_sip_register_service(&session_reinvite_module);
ast_sip_register_service(&outbound_invite_auth_module);
ast_module_shutdown_ref(ast_module_info->self);
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_unregister_service(&outbound_invite_auth_module);
ast_sip_unregister_service(&session_reinvite_module);
ast_sip_unregister_service(&session_module);
ast_sorcery_delete(ast_sip_get_sorcery(), nat_hook);
ao2_cleanup(nat_hook);
ao2_cleanup(sdp_handlers);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "PJSIP Session resource",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
.requires = "res_pjsip",
);
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