mirror of
https://gitlab.gnome.org/GNOME/calls.git
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8b126484cb
Sofia detects a NAT by presence of the "received" parameter in the Via header in
the response to a REGISTER. Sofia will then update the Contact header to use the
IP as reported by the registrar.
The "received" parameter MUST be included in the response according to
https://datatracker.ietf.org/doc/html/rfc3261#section-18.2.1
when the registrar detects a difference between the domain part of the top Via
header and the packet source address but practice has shown that this will not
always be the case.
Addditionally this change allows us to have origins bound to different network
interfaces which would be useful when a registrar can only be accessed through a
VPN.
This also fixes an issue with SDP introduced in
36880c3d34 which was only seen on some SIP
providers:
The session name ("s=") line is not relevant for establishing a connection,
the connection data (c=") line is.
See https://datatracker.ietf.org/doc/html/rfc4566 section 5.3 and 5.7
402 lines
11 KiB
C
402 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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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 (G_OBJECT (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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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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