/* * 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 . * * Author: Evangelos Ribeiro Tzaras * * 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 #include /** * 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; 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 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 (calls_call_get_state (CALLS_CALL (self)) != 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_call_set_state (CALLS_CALL (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 (calls_call_get_state (call)) { 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", calls_call_get_state (call)); } } 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_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, "inbound", inbound, "state", inbound ? CALLS_CALL_STATE_INCOMING : CALLS_CALL_STATE_DIALING, NULL); call->id = g_strdup (id); return call; } /** * 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); }