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asterisk/res/res_pjsip_session.c
Joshua C. Colp 5a5be92b79 bridging: Add better support for adding/removing streams. 
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.

The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.

Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.

ASTERISK-28733

Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5
2020-02-18 10:26:30 -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/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);
}
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;
}
}
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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