1
0
Fork 0
mirror of https://gitlab.gnome.org/GNOME/calls.git synced 2024-11-18 01:25:35 +00:00
Purism-Calls/plugins/sip/calls-sip-call.c
Evangelos Ribeiro Tzaras f206b7d257 call: Rename property from "number" to "id"
The term number is not necessarily accurate when dealing with f.e. SIP.
2021-12-05 09:49:05 +01:00

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 *id;
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_id (CallsCall *call)
{
CallsSipCall *self = CALLS_SIP_CALL (call);
return self->id;
}
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->id);
}
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->id);
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_id = calls_sip_call_get_id;
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 *id,
gboolean inbound,
nua_handle_t *handle)
{
CallsSipCall *call;
g_return_val_if_fail (id != NULL, NULL);
call = g_object_new (CALLS_TYPE_SIP_CALL,
"nua-handle", handle,
NULL);
call->id = g_strdup (id);
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);
}