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