mirror of
https://gitlab.gnome.org/GNOME/calls.git
synced 2024-11-04 23:51:17 +00:00
466 lines
12 KiB
C
466 lines
12 KiB
C
/*
|
|
* Copyright (C) 2021 Purism SPC
|
|
*
|
|
* This file is part of Calls.
|
|
*
|
|
* Calls is free software: you can redistribute it and/or modify it
|
|
* under the terms of the GNU General Public License as published by
|
|
* the Free Software Foundation, either version 3 of the License, or
|
|
* (at your option) any later version.
|
|
*
|
|
* Calls is distributed in the hope that it will be useful, but
|
|
* WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU General Public License
|
|
* along with Calls. If not, see <http://www.gnu.org/licenses/>.
|
|
*
|
|
* Author: Evangelos Ribeiro Tzaras <evangelos.tzaras@puri.sm>
|
|
*
|
|
* SPDX-License-Identifier: GPL-3.0-or-later
|
|
*
|
|
*/
|
|
|
|
#define G_LOG_DOMAIN "CallsSipCall"
|
|
|
|
|
|
#include "calls-call.h"
|
|
#include "calls-message-source.h"
|
|
#include "calls-sip-call.h"
|
|
#include "calls-sip-media-manager.h"
|
|
#include "calls-sip-media-pipeline.h"
|
|
#include "calls-sip-util.h"
|
|
#include "util.h"
|
|
|
|
#include <glib/gi18n.h>
|
|
|
|
#include <sofia-sip/nua.h>
|
|
|
|
/**
|
|
* SECTION:sip-call
|
|
* @short_description: A #CallsCall for the SIP protocol
|
|
* @Title: CallsSipCall
|
|
*
|
|
* #CallsSipCall derives from #CallsCall. Apart from allowing call control
|
|
* like answering and hanging up it also coordinates with #CallsSipMediaManager
|
|
* to prepare and control appropriate #CallsSipMediaPipeline objects.
|
|
*/
|
|
|
|
enum {
|
|
PROP_0,
|
|
PROP_CALL_HANDLE,
|
|
PROP_LAST_PROP
|
|
};
|
|
static GParamSpec *props[PROP_LAST_PROP];
|
|
|
|
struct _CallsSipCall
|
|
{
|
|
GObject parent_instance;
|
|
gchar *number;
|
|
gboolean inbound;
|
|
CallsCallState state;
|
|
|
|
CallsSipMediaManager *manager;
|
|
CallsSipMediaPipeline *pipeline;
|
|
|
|
guint lport_rtp;
|
|
guint lport_rtcp;
|
|
guint rport_rtp;
|
|
guint rport_rtcp;
|
|
gchar *remote;
|
|
|
|
nua_handle_t *nh;
|
|
GList *codecs;
|
|
};
|
|
|
|
static void calls_sip_call_message_source_interface_init (CallsMessageSourceInterface *iface);
|
|
|
|
G_DEFINE_TYPE_WITH_CODE (CallsSipCall, calls_sip_call, CALLS_TYPE_CALL,
|
|
G_IMPLEMENT_INTERFACE (CALLS_TYPE_MESSAGE_SOURCE,
|
|
calls_sip_call_message_source_interface_init))
|
|
|
|
static gboolean
|
|
try_setting_up_media_pipeline (CallsSipCall *self)
|
|
{
|
|
g_assert (CALLS_SIP_CALL (self));
|
|
|
|
if (self->codecs == NULL)
|
|
return FALSE;
|
|
|
|
if (self->pipeline == NULL) {
|
|
MediaCodecInfo *codec = (MediaCodecInfo *) self->codecs->data;
|
|
self->pipeline = calls_sip_media_pipeline_new (codec);
|
|
}
|
|
|
|
if (!self->lport_rtp || !self->lport_rtcp || !self->remote ||
|
|
!self->rport_rtp || !self->rport_rtcp)
|
|
return FALSE;
|
|
|
|
g_debug ("Setting local ports: RTP/RTCP %u/%u",
|
|
self->lport_rtp, self->lport_rtcp);
|
|
|
|
g_object_set (G_OBJECT (self->pipeline),
|
|
"lport-rtp", self->lport_rtp,
|
|
"lport-rtcp", self->lport_rtcp,
|
|
NULL);
|
|
|
|
g_debug ("Setting remote ports: RTP/RTCP %u/%u",
|
|
self->rport_rtp, self->rport_rtcp);
|
|
|
|
g_object_set (G_OBJECT (self->pipeline),
|
|
"remote", self->remote,
|
|
"rport-rtp", self->rport_rtp,
|
|
"rport-rtcp", self->rport_rtcp,
|
|
NULL);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static const char *
|
|
calls_sip_call_get_number (CallsCall *call)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (call);
|
|
|
|
return self->number;
|
|
}
|
|
|
|
|
|
static CallsCallState
|
|
calls_sip_call_get_state (CallsCall *call)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (call);
|
|
|
|
return self->state;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
calls_sip_call_get_inbound (CallsCall *call)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (call);
|
|
|
|
return self->inbound;
|
|
}
|
|
|
|
|
|
static const char *
|
|
calls_sip_call_get_protocol (CallsCall *call)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (call);
|
|
|
|
return get_protocol_from_address (self->number);
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_answer (CallsCall *call)
|
|
{
|
|
CallsSipCall *self;
|
|
g_autofree gchar *local_sdp = NULL;
|
|
guint local_port = get_port_for_rtp ();
|
|
|
|
g_assert (CALLS_IS_CALL (call));
|
|
g_assert (CALLS_IS_SIP_CALL (call));
|
|
|
|
self = CALLS_SIP_CALL (call);
|
|
|
|
g_assert (self->nh);
|
|
|
|
if (self->state != CALLS_CALL_STATE_INCOMING) {
|
|
g_warning ("Call must be in 'incoming' state in order to answer");
|
|
return;
|
|
}
|
|
|
|
/* TODO get free port by creating GSocket and passing that to the pipeline */
|
|
calls_sip_call_setup_local_media_connection (self, local_port, local_port + 1);
|
|
|
|
local_sdp = calls_sip_media_manager_get_capabilities (self->manager,
|
|
local_port,
|
|
FALSE,
|
|
self->codecs);
|
|
|
|
g_assert (local_sdp);
|
|
g_debug ("Setting local SDP to string:\n%s", local_sdp);
|
|
|
|
nua_respond (self->nh, 200, NULL,
|
|
SOATAG_USER_SDP_STR (local_sdp),
|
|
SOATAG_AF (SOA_AF_IP4_IP6),
|
|
TAG_END ());
|
|
|
|
calls_sip_call_set_state (self, CALLS_CALL_STATE_ACTIVE);
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_hang_up (CallsCall *call)
|
|
{
|
|
CallsSipCall *self;
|
|
|
|
g_assert (CALLS_IS_CALL (call));
|
|
g_assert (CALLS_IS_SIP_CALL (call));
|
|
|
|
self = CALLS_SIP_CALL (call);
|
|
|
|
switch (self->state) {
|
|
case CALLS_CALL_STATE_DIALING:
|
|
nua_cancel (self->nh, TAG_END ());
|
|
g_debug ("Hanging up on outgoing ringing call");
|
|
break;
|
|
|
|
case CALLS_CALL_STATE_ACTIVE:
|
|
nua_bye (self->nh, TAG_END ());
|
|
|
|
g_debug ("Hanging up ongoing call");
|
|
break;
|
|
|
|
case CALLS_CALL_STATE_INCOMING:
|
|
nua_respond (self->nh, 480, NULL, TAG_END ());
|
|
g_debug ("Hanging up incoming call");
|
|
break;
|
|
|
|
case CALLS_CALL_STATE_DISCONNECTED:
|
|
g_warning ("Tried hanging up already disconnected call");
|
|
break;
|
|
|
|
default:
|
|
g_warning ("Hanging up not possible in state %d", self->state);
|
|
}
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_set_property (GObject *object,
|
|
guint property_id,
|
|
const GValue *value,
|
|
GParamSpec *pspec)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (object);
|
|
|
|
switch (property_id) {
|
|
case PROP_CALL_HANDLE:
|
|
self->nh = g_value_get_pointer (value);
|
|
break;
|
|
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_get_property (GObject *object,
|
|
guint property_id,
|
|
GValue *value,
|
|
GParamSpec *pspec)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (object);
|
|
|
|
switch (property_id) {
|
|
case PROP_CALL_HANDLE:
|
|
g_value_set_pointer (value, self->nh);
|
|
break;
|
|
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_finalize (GObject *object)
|
|
{
|
|
CallsSipCall *self = CALLS_SIP_CALL (object);
|
|
|
|
g_free (self->number);
|
|
|
|
if (self->pipeline) {
|
|
calls_sip_media_pipeline_stop (self->pipeline);
|
|
g_clear_object (&self->pipeline);
|
|
}
|
|
g_clear_pointer (&self->codecs, g_list_free);
|
|
g_clear_pointer (&self->remote, g_free);
|
|
|
|
G_OBJECT_CLASS (calls_sip_call_parent_class)->finalize (object);
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_class_init (CallsSipCallClass *klass)
|
|
{
|
|
GObjectClass *object_class = G_OBJECT_CLASS (klass);
|
|
CallsCallClass *call_class = CALLS_CALL_CLASS (klass);
|
|
|
|
object_class->get_property = calls_sip_call_get_property;
|
|
object_class->set_property = calls_sip_call_set_property;
|
|
object_class->finalize = calls_sip_call_finalize;
|
|
|
|
call_class->get_number = calls_sip_call_get_number;
|
|
call_class->get_state = calls_sip_call_get_state;
|
|
call_class->get_inbound = calls_sip_call_get_inbound;
|
|
call_class->get_protocol = calls_sip_call_get_protocol;
|
|
call_class->answer = calls_sip_call_answer;
|
|
call_class->hang_up = calls_sip_call_hang_up;
|
|
|
|
props[PROP_CALL_HANDLE] =
|
|
g_param_spec_pointer ("nua-handle",
|
|
"NUA handle",
|
|
"The used NUA handler",
|
|
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY);
|
|
g_object_class_install_property (object_class, PROP_CALL_HANDLE, props[PROP_CALL_HANDLE]);
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_message_source_interface_init (CallsMessageSourceInterface *iface)
|
|
{
|
|
}
|
|
|
|
|
|
static void
|
|
calls_sip_call_init (CallsSipCall *self)
|
|
{
|
|
self->manager = calls_sip_media_manager_default ();
|
|
}
|
|
|
|
/**
|
|
* calls_sip_call_setup_local_media_connection:
|
|
* @self: A #CallsSipCall
|
|
* @port_rtp: The RTP port on the the local host
|
|
* @port_rtcp: The RTCP port on the local host
|
|
*/
|
|
void
|
|
calls_sip_call_setup_local_media_connection (CallsSipCall *self,
|
|
guint port_rtp,
|
|
guint port_rtcp)
|
|
{
|
|
g_return_if_fail (CALLS_IS_SIP_CALL (self));
|
|
|
|
self->lport_rtp = port_rtp;
|
|
self->lport_rtcp = port_rtcp;
|
|
|
|
try_setting_up_media_pipeline (self);
|
|
}
|
|
|
|
/**
|
|
* calls_sip_call_setup_remote_media_connection:
|
|
* @self: A #CallsSipCall
|
|
* @remote: The remote host
|
|
* @port_rtp: The RTP port on the remote host
|
|
* @port_rtcp: The RTCP port on the remote host
|
|
*/
|
|
void
|
|
calls_sip_call_setup_remote_media_connection (CallsSipCall *self,
|
|
const char *remote,
|
|
guint port_rtp,
|
|
guint port_rtcp)
|
|
{
|
|
g_return_if_fail (CALLS_IS_SIP_CALL (self));
|
|
|
|
g_free (self->remote);
|
|
self->remote = g_strdup (remote);
|
|
self->rport_rtp = port_rtp;
|
|
self->rport_rtcp = port_rtcp;
|
|
|
|
try_setting_up_media_pipeline (self);
|
|
}
|
|
|
|
/**
|
|
* calls_sip_call_activate_media:
|
|
* @self: A #CallsSipCall
|
|
* @enabled: %TRUE to enable the media pipeline, %FALSE to disable
|
|
*
|
|
* Controls the state of the #CallsSipMediaPipeline
|
|
*/
|
|
void
|
|
calls_sip_call_activate_media (CallsSipCall *self,
|
|
gboolean enabled)
|
|
{
|
|
g_return_if_fail (CALLS_IS_SIP_CALL (self));
|
|
|
|
/* when hanging up an incoming call the pipeline has not yet been setup */
|
|
if (self->pipeline == NULL && !enabled)
|
|
return;
|
|
g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self->pipeline));
|
|
|
|
if (enabled) {
|
|
calls_sip_media_pipeline_start (self->pipeline);
|
|
} else {
|
|
calls_sip_media_pipeline_stop (self->pipeline);
|
|
}
|
|
}
|
|
|
|
|
|
CallsSipCall *
|
|
calls_sip_call_new (const gchar *number,
|
|
gboolean inbound,
|
|
nua_handle_t *handle)
|
|
{
|
|
CallsSipCall *call;
|
|
|
|
g_return_val_if_fail (number != NULL, NULL);
|
|
|
|
call = g_object_new (CALLS_TYPE_SIP_CALL,
|
|
"nua-handle", handle,
|
|
NULL);
|
|
|
|
call->number = g_strdup (number);
|
|
call->inbound = inbound;
|
|
|
|
if (inbound)
|
|
call->state = CALLS_CALL_STATE_INCOMING;
|
|
else
|
|
call->state = CALLS_CALL_STATE_DIALING;
|
|
|
|
return call;
|
|
}
|
|
|
|
/**
|
|
* calls_sip_call_set_state:
|
|
* @self: A #CallsSipCall
|
|
* @state: The new #CallsCallState to set
|
|
*
|
|
* Sets the new call state and emits the state-changed signal
|
|
*/
|
|
void
|
|
calls_sip_call_set_state (CallsSipCall *self,
|
|
CallsCallState state)
|
|
{
|
|
CallsCallState old_state;
|
|
|
|
g_return_if_fail (CALLS_IS_CALL (self));
|
|
g_return_if_fail (CALLS_IS_SIP_CALL (self));
|
|
|
|
old_state = self->state;
|
|
|
|
if (old_state == state) {
|
|
return;
|
|
}
|
|
|
|
self->state = state;
|
|
g_object_notify (G_OBJECT (self), "state");
|
|
g_signal_emit_by_name (CALLS_CALL (self),
|
|
"state-changed",
|
|
state,
|
|
old_state);
|
|
}
|
|
|
|
/**
|
|
* calls_sip_call_set_codecs:
|
|
* @self: A #CallsSipCall
|
|
* @codecs: A #GList of #MediaCodecInfo elements
|
|
*
|
|
* Set the supported codecs. This is used when answering the call
|
|
*/
|
|
void
|
|
calls_sip_call_set_codecs (CallsSipCall *self,
|
|
GList *codecs)
|
|
{
|
|
g_return_if_fail (CALLS_IS_SIP_CALL (self));
|
|
g_return_if_fail (codecs);
|
|
|
|
g_list_free (self->codecs);
|
|
self->codecs = g_list_copy (codecs);
|
|
}
|