depau/Purism-Calls
1
0
Fork 0
mirror of https://gitlab.gnome.org/GNOME/calls.git synced 2024-12-14 00:17:35 +00:00
Code Issues Releases Wiki Activity
Purism-Calls/plugins/provider/sip/calls-sip-origin.c
Дилян Палаузов 90228e968c SIP dial() convert the port from int to string
2022-11-18 13:48:55 +00:00

1748 lines
51 KiB
C
Raw Blame History

/*
* Copyright (C) 2021-2022 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 "CallsSipOrigin"
#include "calls-account.h"
#include "calls-message-source.h"
#include "calls-origin.h"
#include "calls-sip-call.h"
#include "calls-sip-enums.h"
#include "calls-sip-origin.h"
#include "calls-sip-util.h"
#include "calls-sip-media-manager.h"
#include "calls-network-watch.h"
#include "config.h"
#include "enum-types.h"
#include <glib/gi18n.h>
#include <glib-object.h>
#include <sofia-sip/sofia_features.h>
#include <sofia-sip/nua.h>
#include <sofia-sip/su_tag.h>
#include <sofia-sip/su_tag_io.h>
#include <sofia-sip/sip_util.h>
#include <sofia-sip/sdp.h>
/**
* SECTION:sip-origin
* @short_description: A #CallsOrigin for the SIP protocol
* @Title: CallsSipOrigin
*
* #CallsSipOrigin implements the #CallsOriginInterface and is mainly
* responsible for managing the sofia-sip callbacks, keeping track of #CallsSipCall
* objects and coordinating with #CallsSipMediaManager.
*/
enum {
PROP_0,
PROP_NAME,
PROP_ID,
PROP_ACC_HOST,
PROP_ACC_USER,
PROP_ACC_PASSWORD,
PROP_ACC_DISPLAY_NAME,
PROP_ACC_PORT,
PROP_ACC_PROTOCOL,
PROP_ACC_AUTO_CONNECT,
PROP_ACC_DIRECT,
PROP_ACC_LOCAL_PORT,
PROP_SIP_CONTEXT,
PROP_ACC_STATE,
PROP_ACC_ADDRESS,
PROP_CALLS,
PROP_COUNTRY_CODE,
PROP_CAN_TEL,
PROP_MEDIA_ENCRYPTION,
PROP_LAST_PROP,
};
static GParamSpec *props[PROP_LAST_PROP];
static gboolean set_contact_header = FALSE;
struct _CallsSipOrigin {
GObject parent_instance;
CallsSipContext *ctx;
nua_t *nua;
CallsSipHandles *oper;
char *contact_header; /* Needed for sofia SIP >= 1.13 */
/* Direct connection mode is useful for debugging purposes */
gboolean use_direct_connection;
/* Needed to handle shutdown correctly. See sip_callback and dispose method */
gboolean is_nua_shutdown;
gboolean is_shutdown_success;
CallsAccountState state;
CallsSipMediaManager *media_manager;
/* Account information */
char *host;
char *user;
char *password;
char *display_name;
gint port;
char *transport_protocol;
gboolean auto_connect;
gboolean direct_mode;
gboolean can_tel;
SipMediaEncryption media_encryption;
char *own_ip;
gint local_port;
const char *protocol_prefix;
char *address;
char *name;
char *id;
GList *calls;
GHashTable *call_handles;
};
static void calls_sip_origin_message_source_interface_init (CallsOriginInterface *iface);
static void calls_sip_origin_origin_interface_init (CallsOriginInterface *iface);
static void calls_sip_origin_accounts_interface_init (CallsAccountInterface *iface);
G_DEFINE_TYPE_WITH_CODE (CallsSipOrigin, calls_sip_origin, G_TYPE_OBJECT,
G_IMPLEMENT_INTERFACE (CALLS_TYPE_MESSAGE_SOURCE,
calls_sip_origin_message_source_interface_init)
G_IMPLEMENT_INTERFACE (CALLS_TYPE_ORIGIN,
calls_sip_origin_origin_interface_init)
G_IMPLEMENT_INTERFACE (CALLS_TYPE_ACCOUNT,
calls_sip_origin_accounts_interface_init))
static void
change_state (CallsSipOrigin *self,
CallsAccountState new_state,
CallsAccountStateReason reason)
{
CallsAccountState old_state;
g_assert (CALLS_SIP_ORIGIN (self));
if (self->state == new_state &&
new_state != CALLS_ACCOUNT_STATE_ERROR &&
!calls_account_state_reason_is_for_ui (reason))
return;
old_state = self->state;
self->state = new_state;
g_object_notify_by_pspec (G_OBJECT (self), props[PROP_ACC_STATE]);
g_signal_emit_by_name (self, "account-state-changed", old_state, new_state, reason);
calls_account_emit_message_for_state_change (CALLS_ACCOUNT (self), new_state, reason);
}
static void
remove_call (CallsSipOrigin *self,
CallsCall *call,
const char *reason)
{
CallsOrigin *origin;
CallsSipCall *sip_call;
gboolean inbound;
nua_handle_t *nh;
origin = CALLS_ORIGIN (self);
sip_call = CALLS_SIP_CALL (call);
self->calls = g_list_remove (self->calls, call);
g_object_get (sip_call,
"inbound", &inbound,
"nua-handle", &nh,
NULL);
/* TODO support multiple simultaneous calls */
if (self->oper->call_handle == nh)
self->oper->call_handle = NULL;
g_signal_emit_by_name (origin, "call-removed", call, reason);
g_object_unref (call);
}
static void
remove_calls (CallsSipOrigin *self,
const char *reason)
{
CallsCall *call;
GList *next;
while (self->calls != NULL) {
call = self->calls->data;
next = self->calls->next;
calls_call_hang_up (call);
g_list_free_1 (self->calls);
self->calls = next;
g_signal_emit_by_name (self, "call-removed", call, reason);
g_object_unref (call);
}
g_hash_table_remove_all (self->call_handles);
g_clear_pointer (&self->oper->call_handle, nua_handle_unref);
}
static void
on_call_state_changed (CallsSipCall *call,
GParamSpec *pspec,
CallsSipOrigin *self)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
g_assert (CALLS_IS_CALL (call));
if (calls_call_get_state (CALLS_CALL (call)) != CALLS_CALL_STATE_DISCONNECTED)
return;
remove_call (self, CALLS_CALL (call), "Disconnected");
}
static void
add_call (CallsSipOrigin *self,
const char *address,
gboolean inbound,
nua_handle_t *handle)
{
CallsSipCall *sip_call;
CallsCall *call;
CallsSipMediaPipeline *pipeline;
const char *call_address = address;
gint rtp_port, rtcp_port;
g_auto (GStrv) address_split = NULL;
g_autofree char *local_sdp = NULL;
pipeline = calls_sip_media_manager_get_pipeline (self->media_manager);
rtp_port = calls_sip_media_pipeline_get_rtp_port (pipeline);
rtcp_port = calls_sip_media_pipeline_get_rtcp_port (pipeline);
if (self->can_tel) {
address_split = g_strsplit_set (address, ":@;", -1);
if (g_strv_length (address_split) >=2 &&
g_strcmp0 (address_split[2], self->host) == 0)
call_address = address_split[1];
}
sip_call = calls_sip_call_new (call_address,
inbound,
self->own_ip,
pipeline,
self->media_encryption,
handle);
g_assert (sip_call != NULL);
if (self->oper->call_handle)
nua_handle_unref (self->oper->call_handle);
self->oper->call_handle = handle;
self->calls = g_list_append (self->calls, sip_call);
g_hash_table_insert (self->call_handles, handle, sip_call);
call = CALLS_CALL (sip_call);
g_signal_emit_by_name (CALLS_ORIGIN (self), "call-added", call);
g_signal_connect (call, "notify::state",
G_CALLBACK (on_call_state_changed),
self);
if (!inbound) {
g_autoptr (GList) crypto_attributes = NULL;
g_autoptr (CallsSdpCryptoContext) ctx =
calls_sip_call_get_sdp_crypto_context (sip_call);
if (self->media_encryption == SIP_MEDIA_ENCRYPTION_FORCED) {
if (!calls_sdp_crypto_context_generate_offer (ctx)) {
g_warning ("Media encryption must be used, but could not generate offer. Aborting");
calls_call_set_state (CALLS_CALL (sip_call), CALLS_CALL_STATE_DISCONNECTED);
return;
}
}
if (self->media_encryption == SIP_MEDIA_ENCRYPTION_PREFERRED) {
if (!calls_sdp_crypto_context_generate_offer (ctx))
g_debug ("Media encryption optional, but could not generate offer. Continuing unencrypted");
}
crypto_attributes = calls_sdp_crypto_context_get_crypto_candidates (ctx, FALSE);
local_sdp = calls_sip_media_manager_static_capabilities (self->media_manager,
self->own_ip,
rtp_port,
rtcp_port,
crypto_attributes);
g_assert (local_sdp);
g_debug ("Setting local SDP for outgoing call to %s:\n%s", address, local_sdp);
/* TODO handle IPv4 vs IPv6 for nua_invite (SOATAG_TAG) */
nua_invite (self->oper->call_handle,
SOATAG_AF (SOA_AF_IP4_IP6),
SOATAG_USER_SDP_STR (local_sdp),
SIPTAG_TO_STR (address),
TAG_IF (!!self->contact_header, SIPTAG_CONTACT_STR (self->contact_header)),
SOATAG_RTP_SORT (SOA_RTP_SORT_REMOTE),
SOATAG_RTP_SELECT (SOA_RTP_SELECT_ALL),
TAG_END ());
}
}
static void
dial (CallsOrigin *origin,
const char *address)
{
CallsSipOrigin *self;
nua_handle_t *nh;
g_autofree char *name = NULL;
g_autofree char *dial_target = NULL;
g_assert (CALLS_ORIGIN (origin));
g_assert (CALLS_IS_SIP_ORIGIN (origin));
name = calls_origin_get_name (origin);
if (!address || !*address) {
g_warning ("Tried dialing on origin '%s' without an address",
name);
return;
}
if (calls_account_get_state (CALLS_ACCOUNT (origin)) !=
CALLS_ACCOUNT_STATE_ONLINE) {
g_warning ("Tried dialing '%s' on origin '%s', but it's not online",
address, name);
return;
}
self = CALLS_SIP_ORIGIN (origin);
nh = nua_handle (self->nua, self->oper,
NUTAG_MEDIA_ENABLE (1),
SOATAG_ACTIVE_AUDIO (SOA_ACTIVE_SENDRECV),
TAG_END ());
/* Make sure @host is in the dial target */
if (g_strstr_len (address, -1, "@")) {
dial_target = g_strdup (address);
} else {
if (self->port > 0)
dial_target = g_strdup_printf ("%s@%s:%d", address, self->host, self->port);
else
dial_target = g_strconcat (address, "@", self->host, NULL);
}
g_debug ("Calling `%s' from origin '%s'", address, name);
/* We don't require the user to input the prefix */
if (!g_str_has_prefix (address, "sip:") &&
!g_str_has_prefix (address, "sips:")) {
g_autofree char * target_with_prefix =
g_strconcat (self->protocol_prefix, ":", dial_target, NULL);
add_call (CALLS_SIP_ORIGIN (origin), target_with_prefix, FALSE, nh);
} else {
add_call (CALLS_SIP_ORIGIN (origin), dial_target, FALSE, nh);
}
}
static void
create_inbound (CallsSipOrigin *self,
const char *address,
nua_handle_t *handle)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
g_assert (address != NULL);
/* TODO support multiple calls */
if (self->oper->call_handle)
nua_handle_unref (self->oper->call_handle);
self->oper->call_handle = handle;
add_call (self, address, TRUE, handle);
}
static void
sip_authenticate (CallsSipOrigin *self,
nua_handle_t *nh,
sip_t const *sip)
{
const char *scheme = NULL;
const char *realm = NULL;
g_autofree char *auth = NULL;
sip_www_authenticate_t *www_auth = sip->sip_www_authenticate;
sip_proxy_authenticate_t *proxy_auth = sip->sip_proxy_authenticate;
if (www_auth) {
scheme = www_auth->au_scheme;
realm = msg_params_find (www_auth->au_params, "realm=");
} else if (proxy_auth) {
scheme = proxy_auth->au_scheme;
realm = msg_params_find (proxy_auth->au_params, "realm=");
} else {
g_warning ("No authentication context found");
return;
}
g_debug ("need to authenticate to realm %s", realm);
/* TODO handle authentication to different realms
* https://source.puri.sm/Librem5/calls/-/issues/266
*/
auth = g_strdup_printf ("%s:%s:%s:%s",
scheme, realm, self->user, self->password);
nua_authenticate (nh, NUTAG_AUTH (auth), TAG_END ());
}
static void
sip_r_invite (int status,
char const *phrase,
nua_t *nua,
CallsSipOrigin *origin,
nua_handle_t *nh,
CallsSipHandles *op,
sip_t const *sip,
tagi_t tags[])
{
g_debug ("response to outgoing INVITE: %03d %s", status, phrase);
/* TODO call states (see i_state) */
if (status == 401 || status == 407) {
sip_authenticate (origin, nh, sip);
} else if (status == 403) {
g_warning ("Response to outgoing INVITE: 403 wrong credentials?");
} else if (status == 904) {
g_warning ("Response to outgoing INVITE: 904 unmatched challenge."
"Possibly the challenge was already answered?");
} else if (status == 180) {
} else if (status == 100) {
} else if (status == 200) {
}
}
static void
sip_r_register (int status,
char const *phrase,
nua_t *nua,
CallsSipOrigin *origin,
nua_handle_t *nh,
CallsSipHandles *op,
sip_t const *sip,
tagi_t tags[])
{
if (status == 200) {
if (origin->state == CALLS_ACCOUNT_STATE_ONLINE)
return;
g_debug ("REGISTER successful");
change_state (origin,
CALLS_ACCOUNT_STATE_ONLINE,
CALLS_ACCOUNT_STATE_REASON_CONNECTED);
/* Fix for https://github.com/freeswitch/sofia-sip/issues/25 */
if (!origin->contact_header)
nua_get_params (nua, TAG_ANY (), TAG_END ());
if (sip->sip_contact && sip->sip_contact->m_url->url_host) {
if (g_strcmp0 (origin->own_ip, sip->sip_contact->m_url->url_host) == 0)
return;
g_free (origin->own_ip);
origin->own_ip = g_strdup (sip->sip_contact->m_url->url_host);
g_debug ("Own IP as reported by the registrar: %s", origin->own_ip);
}
} else if (status == 100) {
/* nothing to do; request authorized by cache */
} else if (status == 401 || status == 407) {
sip_authenticate (origin, nh, sip);
} else if (status == 403) {
g_debug ("REGISTER denied: Probably using wrong credentials");
change_state (origin,
CALLS_ACCOUNT_STATE_OFFLINE,
CALLS_ACCOUNT_STATE_REASON_AUTHENTICATION_FAILURE);
} else if (status == 503) {
change_state (origin,
CALLS_ACCOUNT_STATE_OFFLINE,
CALLS_ACCOUNT_STATE_REASON_CONNECTION_DNS_ERROR);
} else if (status == 904) {
g_warning ("REGISTER: unmatched challenge");
change_state (origin,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_INTERNAL_ERROR);
} else {
g_warning ("REGISTER: %d %s", status, phrase);
change_state (origin,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_INTERNAL_ERROR);
}
}
static void
sip_r_unregister (int status,
char const *phrase,
nua_t *nua,
CallsSipOrigin *origin,
nua_handle_t *nh,
CallsSipHandles *op,
sip_t const *sip,
tagi_t tags[])
{
g_debug ("response to unregistering: %03d %s", status, phrase);
if (status == 200) {
g_debug ("Unregistering successful");
change_state (origin,
CALLS_ACCOUNT_STATE_OFFLINE,
CALLS_ACCOUNT_STATE_REASON_DISCONNECTED);
} else if (status == 100) {
/* nothing to do; request authorized by cache */
} else if (status == 503) {
change_state (origin,
CALLS_ACCOUNT_STATE_OFFLINE,
CALLS_ACCOUNT_STATE_REASON_CONNECTION_DNS_ERROR);
} else {
g_warning ("Unregisterung unsuccessful: %03d %s", status, phrase);
change_state (origin,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_UNKNOWN);
}
}
static void
sip_i_state (int status,
char const *phrase,
nua_t *nua,
CallsSipOrigin *origin,
nua_handle_t *nh,
CallsSipHandles *op,
sip_t const *sip,
tagi_t tags[])
{
const sdp_session_t *r_sdp = NULL;
const sdp_session_t *l_sdp = NULL;
const char *r_sdp_str = NULL;
gint call_state = nua_callstate_init;
CallsCallState state;
CallsSipCall *call;
int offer_sent = 0;
int offer_recv = 0;
int answer_sent = 0;
int answer_recv = 0;
g_assert (CALLS_IS_SIP_ORIGIN (origin));
g_debug ("The call state has changed: %03d %s", status, phrase);
call = g_hash_table_lookup (origin->call_handles, nh);
if (call == NULL) {
g_warning ("No call found for the current handle");
return;
}
tl_gets (tags,
SOATAG_LOCAL_SDP_REF (l_sdp),
SOATAG_REMOTE_SDP_REF (r_sdp),
SOATAG_REMOTE_SDP_STR_REF (r_sdp_str),
NUTAG_CALLSTATE_REF (call_state),
NUTAG_OFFER_SENT_REF (offer_sent),
NUTAG_OFFER_RECV_REF (offer_recv),
NUTAG_ANSWER_SENT_REF (answer_sent),
NUTAG_ANSWER_RECV_REF (answer_recv),
TAG_END ());
if (status == 503) {
CALLS_EMIT_MESSAGE (origin, "DNS error", GTK_MESSAGE_ERROR);
}
if (r_sdp) {
g_autoptr (CallsSdpCryptoContext) ctx = NULL;
g_autoptr (GList) remote_codecs =
calls_sip_media_manager_get_codecs_from_sdp (origin->media_manager,
r_sdp->sdp_media);
g_autoptr (GList) codecs = NULL;
GList *node;
GList *preferred_codecs = calls_sip_media_manager_codec_candidates (origin->media_manager);
const char *session_ip = NULL;
const char *media_ip = NULL;
int rtp_port;
int rtcp_port = 0;
for (node = remote_codecs; node != NULL; node = node->next) {
MediaCodecInfo *codec = node->data;
if (g_list_find (preferred_codecs, codec)) {
g_debug ("Found common codec: %s", codec->name);
codecs = g_list_append (codecs, codec);
}
}
g_debug ("Remote SDP was set to:\n%s", r_sdp_str);
if (!codecs) {
g_warning ("No common codecs in SDP. Hanging up. Remote SDP:\n%s", r_sdp_str);
calls_call_hang_up (CALLS_CALL (call));
return;
}
ctx = calls_sip_call_get_sdp_crypto_context (call);
if (origin->media_encryption == SIP_MEDIA_ENCRYPTION_FORCED) {
if (!calls_sdp_crypto_context_set_remote_media (ctx, r_sdp->sdp_media)) {
g_warning ("Media encryption is enforced, but remote didn't set crypto attributes.\n"
"Remote SDP: %s", r_sdp_str);
calls_call_hang_up (CALLS_CALL (call));
return;
}
}
if (origin->media_encryption == SIP_MEDIA_ENCRYPTION_PREFERRED) {
if (!calls_sdp_crypto_context_set_remote_media (ctx, r_sdp->sdp_media))
g_debug ("Remote didn't set crypto attributes. Continuing unencrypted");
}
if (origin->media_encryption == SIP_MEDIA_ENCRYPTION_NONE) {
gboolean got_crypto = FALSE;
for (sdp_attribute_t *attr = r_sdp->sdp_media->m_attributes; attr; attr = attr->a_next) {
if (g_strcmp0 (attr->a_name, "crypto") == 0) {
got_crypto = TRUE;
break;
}
}
if (got_crypto) {
g_debug ("Remote peer %s wants to talk using encryption, but not allowed by local policy",
calls_call_get_id (CALLS_CALL (call)));
calls_call_hang_up (CALLS_CALL (call));
return;
}
}
if (r_sdp->sdp_connection && r_sdp->sdp_connection->c_address)
session_ip = r_sdp->sdp_connection->c_address;
if (r_sdp->sdp_media && r_sdp->sdp_media->m_connections &&
r_sdp->sdp_media->m_connections->c_address)
media_ip = r_sdp->sdp_media->m_connections->c_address;
if (!session_ip && !media_ip) {
g_warning ("Could not determine IP of remote peer. Hanging up");
calls_call_hang_up (CALLS_CALL (call));
return;
}
calls_sip_call_set_codecs (call, codecs);
/* TODO This needs to adapt for the ICE case */
rtp_port = r_sdp->sdp_media->m_port;
for (sdp_attribute_t *attr = r_sdp->sdp_media->m_attributes; attr; attr = attr->a_next) {
if (g_strcmp0 (attr->a_name, "rtcp") == 0) {
rtcp_port = atoi (attr->a_value);
break;
}
}
/* Legacy fallback */
if (rtcp_port == 0)
rtcp_port = rtp_port + 1;
calls_sip_call_setup_remote_media_connection (call,
media_ip ?: session_ip,
rtp_port,
rtcp_port);
}
switch (call_state) {
case nua_callstate_init:
return;
case nua_callstate_calling:
state = CALLS_CALL_STATE_DIALING;
break;
case nua_callstate_received:
state = CALLS_CALL_STATE_INCOMING;
g_debug ("Call incoming");
break;
case nua_callstate_ready:
g_debug ("Call ready. Activating media pipeline");
calls_sip_call_activate_media (call, TRUE);
state = CALLS_CALL_STATE_ACTIVE;
break;
case nua_callstate_terminated:
g_debug ("Call terminated. Deactivating media pipeline");
calls_sip_call_activate_media (call, FALSE);
state = CALLS_CALL_STATE_DISCONNECTED;
g_hash_table_remove (origin->call_handles, nh);
break;
default:
return;
}
calls_call_set_state (CALLS_CALL (call), state);
}
static void
sip_r_get_params (int status,
char const *phrase,
nua_t *nua,
CallsSipOrigin *origin,
nua_handle_t *nh,
CallsSipHandles *op,
sip_t const *sip,
tagi_t tags[])
{
tagi_t *from = NULL;
const char *from_str = NULL;
g_debug ("response to get_params: %03d %s", status, phrase);
if (!set_contact_header)
return;
if ((status != 200) || ((from = tl_find (tags, siptag_from_str)) == NULL)) {
g_warning ("Could not find 'siptag_from_tag' among the tags");
return;
}
from_str = (const char *) from->t_value;
g_free (origin->contact_header);
origin->contact_header = g_strdup (from_str);
}
static void
sip_callback (nua_event_t event,
int status,
char const *phrase,
nua_t *nua,
nua_magic_t *magic,
nua_handle_t *nh,
nua_hmagic_t *hmagic,
sip_t const *sip,
tagi_t tags[])
{
CallsSipOrigin *origin = CALLS_SIP_ORIGIN (magic);
CallsSipHandles *op = origin->oper;
g_autofree char *from = NULL;
switch (event) {
case nua_i_invite:
if (sip->sip_from &&
sip->sip_from->a_url->url_scheme &&
sip->sip_from->a_url->url_user &&
sip->sip_from->a_url->url_host)
from = g_strconcat (sip->sip_from->a_url->url_scheme, ":",
sip->sip_from->a_url->url_user, "@",
sip->sip_from->a_url->url_host, NULL);
else {
nua_respond (nh, 400, NULL, TAG_END ());
g_warning ("invalid incoming INVITE request");
break;
}
g_debug ("incoming call INVITE: %03d %s from %s", status, phrase, from);
/* reject if there already is a ongoing call */
if (op->call_handle) {
nua_respond (nh, 486, NULL, TAG_END ());
g_debug ("Cannot handle more than one call. Rejecting");
} else {
create_inbound (origin, from, nh);
}
break;
case nua_r_invite:
sip_r_invite (status,
phrase,
nua,
origin,
nh,
op,
sip,
tags);
break;
case nua_i_ack:
g_debug ("incoming ACK: %03d %s", status, phrase);
break;
case nua_i_bye:
g_debug ("incoming BYE: %03d %s", status, phrase);
break;
case nua_r_bye:
g_debug ("response to BYE: %03d %s", status, phrase);
if (status == 200) {
CallsSipCall *call =
CALLS_SIP_CALL (g_hash_table_lookup (origin->call_handles, nh));
if (call == NULL) {
g_warning ("BYE response from an unknown call");
return;
}
}
break;
case nua_r_register:
sip_r_register (status,
phrase,
nua,
origin,
nh,
op,
sip,
tags);
break;
case nua_r_unregister:
sip_r_unregister (status,
phrase,
nua,
origin,
nh,
op,
sip,
tags);
break;
case nua_r_get_params:
sip_r_get_params (status,
phrase,
nua,
origin,
nh,
op,
sip,
tags);
break;
case nua_r_set_params:
g_debug ("response to set_params: %03d %s", status, phrase);
break;
case nua_i_outbound:
g_debug ("status of outbound engine has changed: %03d %s", status, phrase);
if (status == 404) {
CALLS_EMIT_MESSAGE (origin, "contact not found", GTK_MESSAGE_ERROR);
g_warning ("outbound engine changed state to %03d %s", status, phrase);
}
break;
case nua_i_state:
sip_i_state (status,
phrase,
nua,
origin,
nh,
op,
sip,
tags);
break;
case nua_r_cancel:
g_debug ("response to CANCEL: %03d %s", status, phrase);
break;
case nua_r_terminate:
break;
case nua_r_shutdown:
/* see also deinit_sip () */
g_debug ("response to nua_shutdown: %03d %s", status, phrase);
if (status == 200) {
origin->is_nua_shutdown = TRUE;
origin->is_shutdown_success = TRUE;
} else if (status == 500) {
origin->is_nua_shutdown = TRUE;
origin->is_shutdown_success = FALSE;
}
break;
/* Deprecated events */
case nua_i_active:
break;
case nua_i_terminated:
break;
default:
/* unknown event */
g_debug ("unknown event %d: %03d %s",
event,
status,
phrase);
if (tags && tags[0].t_tag != NULL) {
g_debug ("printing tags");
tl_print (stdout, "", tags);
}
break;
}
}
static nua_t *
setup_nua (CallsSipOrigin *self)
{
const char *sip_test_env = g_getenv ("CALLS_SIP_TEST");
nua_t *nua;
gboolean use_sips = FALSE;
gboolean use_ipv6 = FALSE; /* TODO make configurable or use DNS to figure out if ipv6 is supported*/
const char *ipv6_bind = "*";
const char *ipv4_bind = "0.0.0.0";
const char *uuid = NULL;
g_autofree char *urn_uuid = NULL;
g_autofree char *sip_url = NULL;
g_autofree char *sips_url = NULL;
g_autofree char *from_str = NULL;
if (!sip_test_env || sip_test_env[0] == '\0') {
CallsNetworkWatch *nw = calls_network_watch_get_default ();
if (nw) {
ipv4_bind = calls_network_watch_get_ipv4 (nw);
ipv6_bind = calls_network_watch_get_ipv6 (nw);
}
}
uuid = nua_generate_instance_identifier (self->ctx->home);
urn_uuid = g_strdup_printf ("urn:uuid:%s", uuid);
self->protocol_prefix = get_protocol_prefix (self->transport_protocol);
self->address = g_strconcat (self->user, "@", self->host, NULL);
from_str = g_strconcat (self->protocol_prefix, ":", self->address, NULL);
g_object_notify_by_pspec (G_OBJECT (self), props[PROP_ACC_ADDRESS]);
use_sips = check_sips (from_str);
use_ipv6 = check_ipv6 (self->host);
if (self->local_port > 0) {
sip_url = g_strdup_printf ("sip:%s:%d",
use_ipv6 ? ipv6_bind : ipv4_bind,
self->local_port);
sips_url = g_strdup_printf ("sips:%s:%d",
use_ipv6 ? ipv6_bind : ipv4_bind,
self->local_port);
} else {
sip_url = g_strdup_printf ("sip:%s:*",
use_ipv6 ? ipv6_bind : ipv4_bind);
sips_url = g_strdup_printf ("sips:%s:*",
use_ipv6 ? ipv6_bind : ipv4_bind);
}
/** For TLS nua_create() will error if NUTAG_URL includes ";transport=tls"
* In that case NUTAG_SIPS_URL should be used and NUTAG_URL should be as usual
* Since UDP is the default we only need to append the suffix in the TCP case
*/
if (g_ascii_strcasecmp (self->transport_protocol, "TCP") == 0) {
char *temp = sip_url;
sip_url = g_strdup_printf ("%s;transport=%s", temp, self->transport_protocol);
g_free (temp);
}
nua = nua_create (self->ctx->root,
sip_callback,
self,
NUTAG_USER_AGENT (APP_DATA_NAME),
NUTAG_URL (sip_url),
TAG_IF (use_sips, NUTAG_SIPS_URL (sips_url)),
SIPTAG_FROM_STR (from_str),
NUTAG_ALLOW ("INVITE, ACK, BYE, CANCEL, OPTIONS, UPDATE"),
NUTAG_SUPPORTED ("replaces, gruu, outbound"),
NTATAG_MAX_FORWARDS (70),
NUTAG_ENABLEINVITE (1),
NUTAG_AUTOANSWER (0),
NUTAG_AUTOACK (1),
NUTAG_PATH_ENABLE (0),
NUTAG_MEDIA_ENABLE (1),
NUTAG_INSTANCE (urn_uuid),
TAG_NULL ());
return nua;
}
static char *
get_registrar_url (CallsSipOrigin *self)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
if (self->port > 0 && self->port <= 65535)
return g_strdup_printf ("%s:%s:%d", self->protocol_prefix, self->host, self->port);
else
return g_strconcat (self->protocol_prefix, ":", self->host, NULL);
}
static CallsSipHandles *
setup_sip_handles (CallsSipOrigin *self)
{
CallsSipHandles *oper;
g_autofree char *registrar_url = NULL;
g_assert (CALLS_IS_SIP_ORIGIN (self));
if (!(oper = su_zalloc (self->ctx->home, sizeof(CallsSipHandles)))) {
g_warning ("cannot create handle");
return NULL;
}
oper->context = self->ctx;
oper->register_handle = nua_handle (self->nua, self->oper,
SIPTAG_EXPIRES_STR ("180"),
NUTAG_SUPPORTED ("replaces, outbound, gruu"),
NUTAG_OUTBOUND ("outbound natify gruuize validate"), // <- janus uses "no-validate"
NUTAG_M_PARAMS ("user=phone"),
NUTAG_CALLEE_CAPS (1), /* header parameters based on SDP capabilities and Allow header */
TAG_END ());
oper->call_handle = NULL;
return oper;
}
static void
go_online (CallsAccount *account,
gboolean online)
{
CallsSipOrigin *self;
g_assert (CALLS_IS_ACCOUNT (account));
g_assert (CALLS_IS_SIP_ORIGIN (account));
self = CALLS_SIP_ORIGIN (account);
if (self->use_direct_connection)
return;
if (!self->nua) {
g_warning ("Cannot go online: nua handle not initialized");
g_object_notify_by_pspec (G_OBJECT (self), props[PROP_ACC_STATE]);
return;
}
if (online) {
g_autofree char *registrar_url = NULL;
if (self->state == CALLS_ACCOUNT_STATE_ONLINE)
return;
registrar_url = get_registrar_url (self);
nua_register (self->oper->register_handle,
NUTAG_M_USERNAME (self->user),
TAG_IF (self->display_name, NUTAG_M_DISPLAY (self->display_name)),
NUTAG_REGISTRAR (registrar_url),
TAG_END ());
} else {
if (self->state == CALLS_ACCOUNT_STATE_OFFLINE)
return;
nua_unregister (self->oper->register_handle,
TAG_END ());
}
}
static const char *
get_address (CallsAccount *account)
{
CallsSipOrigin *self;
g_assert (CALLS_IS_ACCOUNT (account));
g_assert (CALLS_IS_SIP_ORIGIN (account));
self = CALLS_SIP_ORIGIN (account);
return self->address;
}
static void
setup_account_for_direct_connection (CallsSipOrigin *self)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
/* honour username, if previously set */
if (self->user == NULL)
self->user = g_strdup (g_get_user_name ());
g_free (self->host);
self->host = g_strdup (g_get_host_name ());
g_clear_pointer (&self->password, g_free);
g_free (self->transport_protocol);
self->transport_protocol = g_strdup ("UDP");
self->protocol_prefix = get_protocol_prefix ("UDP");
g_debug ("Account changed:\nuser: %s\nhost: %s",
self->user, self->host);
}
static gboolean
is_account_complete (CallsSipOrigin *self)
{
/* we need only need to check for password if needing to authenticate over a proxy/UAS */
if (self->user == NULL ||
(!self->use_direct_connection && self->password == NULL) ||
self->host == NULL ||
self->transport_protocol == NULL)
return FALSE;
return TRUE;
}
static gboolean
init_sip_account (CallsSipOrigin *self,
GError **error)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
change_state (self,
CALLS_ACCOUNT_STATE_INITIALIZING,
CALLS_ACCOUNT_STATE_REASON_INITIALIZATION_STARTED);
if (self->use_direct_connection) {
g_debug ("Direct connection case. Using user and hostname");
setup_account_for_direct_connection (self);
}
if (!is_account_complete (self)) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Must have completed account setup before calling "
"init_sip_account (). "
"Try again when account is setup");
change_state (self,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_NO_CREDENTIALS);
return FALSE;
}
/* setup_nua() and setup_sip_handles() only after account data has been set */
self->nua = setup_nua (self);
if (self->nua == NULL) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed setting up nua context");
change_state (self,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_INTERNAL_ERROR);
return FALSE;
}
self->oper = setup_sip_handles (self);
if (self->oper == NULL) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
"Failed setting operation handles");
change_state (self,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_INTERNAL_ERROR);
return FALSE;
}
/* In the case of a direct connection we're immediately good to go */
change_state (self,
self->use_direct_connection ? CALLS_ACCOUNT_STATE_ONLINE : CALLS_ACCOUNT_STATE_OFFLINE,
CALLS_ACCOUNT_STATE_REASON_INITIALIZED);
if (self->auto_connect)
go_online (CALLS_ACCOUNT (self), TRUE);
return TRUE;
}
static gboolean
deinit_sip_account (CallsSipOrigin *self)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
if (self->state == CALLS_ACCOUNT_STATE_UNKNOWN)
return TRUE;
change_state (self,
CALLS_ACCOUNT_STATE_DEINITIALIZING,
CALLS_ACCOUNT_STATE_REASON_DEINITIALIZATION_STARTED);
remove_calls (self, NULL);
if (self->nua) {
g_debug ("Clearing any handles");
g_clear_pointer (&self->oper->register_handle, nua_handle_destroy);
g_debug ("Requesting nua_shutdown ()");
self->is_nua_shutdown = FALSE;
self->is_shutdown_success = FALSE;
nua_shutdown (self->nua);
// need to wait for nua_r_shutdown event before calling nua_destroy ()
while (!self->is_nua_shutdown)
su_root_step (self->ctx->root, 100);
if (!self->is_shutdown_success) {
g_warning ("nua_shutdown() timed out. Cannot proceed");
change_state (self,
CALLS_ACCOUNT_STATE_ERROR,
CALLS_ACCOUNT_STATE_REASON_INTERNAL_ERROR);
return FALSE;
}
g_debug ("nua_shutdown() complete. Destroying nua handle");
nua_destroy (self->nua);
self->nua = NULL;
}
g_clear_pointer (&self->own_ip, g_free);
change_state (self,
CALLS_ACCOUNT_STATE_UNKNOWN,
CALLS_ACCOUNT_STATE_REASON_DEINITIALIZED);
return TRUE;
}
static void
recreate_sip (CallsSipOrigin *self)
{
if (deinit_sip_account (self))
init_sip_account (self, NULL);
}
static gboolean
supports_protocol (CallsOrigin *origin,
const char *protocol)
{
CallsSipOrigin *self;
g_assert (protocol);
g_assert (CALLS_IS_SIP_ORIGIN (origin));
self = CALLS_SIP_ORIGIN (origin);
if (g_strcmp0 (protocol, "sip") == 0)
return TRUE;
if (g_strcmp0 (protocol, "sips") == 0)
return g_strcmp0 (self->protocol_prefix, "sips") == 0;
if (g_strcmp0 (protocol, "tel") == 0)
return self->can_tel;
return FALSE;
}
static void
update_name (CallsSipOrigin *self)
{
g_assert (CALLS_IS_SIP_ORIGIN (self));
if (self->display_name && self->display_name[0] != '\0')
self->name = self->display_name;
else
self->name = self->user;
g_object_notify_by_pspec (G_OBJECT (self), props[PROP_NAME]);
}
static void
calls_sip_origin_set_property (GObject *object,
guint property_id,
const GValue *value,
GParamSpec *pspec)
{
CallsSipOrigin *self = CALLS_SIP_ORIGIN (object);
switch (property_id) {
case PROP_ID: /* contruct only */
self->id = g_value_dup_string (value);
break;
case PROP_ACC_HOST:
g_free (self->host);
self->host = g_value_dup_string (value);
break;
case PROP_ACC_DISPLAY_NAME:
g_free (self->display_name);
self->display_name = g_value_dup_string (value);
break;
case PROP_ACC_USER:
g_free (self->user);
self->user = g_value_dup_string (value);
break;
case PROP_ACC_PASSWORD:
g_free (self->password);
self->password = g_value_dup_string (value);
break;
case PROP_ACC_PORT:
self->port = g_value_get_int (value);
break;
case PROP_ACC_PROTOCOL:
g_free (self->transport_protocol);
self->transport_protocol = g_value_dup_string (value);
break;
case PROP_ACC_AUTO_CONNECT:
self->auto_connect = g_value_get_boolean (value);
break;
case PROP_ACC_DIRECT:
self->use_direct_connection = g_value_get_boolean (value);
break;
case PROP_SIP_CONTEXT:
self->ctx = g_value_get_pointer (value);
break;
case PROP_ACC_LOCAL_PORT:
if (g_value_get_int (value) > 0 && g_value_get_int (value) < 1025) {
g_warning ("Tried setting a privileged port as the local port to bind to: %d\n"
"Continue using old 'local-port' value: %d (using 0 let's the OS decide)",
g_value_get_int (value), self->local_port);
return;
}
self->local_port = g_value_get_int (value);
break;
case PROP_CAN_TEL:
self->can_tel = g_value_get_boolean (value);
break;
case PROP_MEDIA_ENCRYPTION:
self->media_encryption = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
calls_sip_origin_get_property (GObject *object,
guint property_id,
GValue *value,
GParamSpec *pspec)
{
CallsSipOrigin *self = CALLS_SIP_ORIGIN (object);
switch (property_id) {
case PROP_NAME:
g_value_set_string (value, self->name);
break;
case PROP_ID:
g_value_set_string (value, self->id);
break;
case PROP_ACC_HOST:
g_value_set_string (value, self->host);
break;
case PROP_ACC_DISPLAY_NAME:
g_value_set_string (value, self->display_name);
break;
case PROP_ACC_USER:
g_value_set_string (value, self->user);
break;
case PROP_ACC_PASSWORD:
g_value_set_string (value, self->password);
break;
case PROP_ACC_PORT:
g_value_set_int (value, self->port);
break;
case PROP_ACC_PROTOCOL:
g_value_set_string (value, self->transport_protocol);
break;
case PROP_ACC_AUTO_CONNECT:
g_value_set_boolean (value, self->auto_connect);
break;
case PROP_ACC_DIRECT:
g_value_set_boolean (value, self->direct_mode);
break;
case PROP_ACC_LOCAL_PORT:
g_value_set_int (value, self->local_port);
break;
case PROP_CALLS:
g_value_set_pointer (value, g_list_copy (self->calls));
break;
case PROP_ACC_STATE:
g_value_set_enum (value, self->state);
break;
case PROP_ACC_ADDRESS:
g_value_set_string (value, get_address (CALLS_ACCOUNT (self)));
break;
case PROP_COUNTRY_CODE:
g_value_set_string (value, NULL);
break;
case PROP_CAN_TEL:
g_value_set_boolean (value, self->can_tel);
break;
case PROP_MEDIA_ENCRYPTION:
g_value_set_enum (value, self->media_encryption);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
calls_sip_origin_constructed (GObject *object)
{
CallsSipOrigin *self = CALLS_SIP_ORIGIN (object);
g_autoptr (GError) error = NULL;
int major = 0;
int minor = 0;
int patch = 0;
if (sscanf (SOFIA_SIP_VERSION, "%d.%d.%d", &major, &minor, &patch) == 3) {
if (major > 2 || (major == 1 && minor >= 13)) {
/* Sofia 1.13 does no longer include the contact header by default, see #304 */
set_contact_header = TRUE;
}
}
if (!init_sip_account (self, &error)) {
g_warning ("Error initializing the SIP account: %s", error->message);
}
update_name (self);
self->media_manager = calls_sip_media_manager_default ();
G_OBJECT_CLASS (calls_sip_origin_parent_class)->constructed (object);
}
static void
calls_sip_origin_dispose (GObject *object)
{
CallsSipOrigin *self = CALLS_SIP_ORIGIN (object);
g_clear_pointer (&self->id, g_free);
g_clear_pointer (&self->own_ip, g_free);
g_clear_pointer (&self->transport_protocol, g_free);
g_clear_pointer (&self->display_name, g_free);
g_clear_pointer (&self->host, g_free);
g_clear_pointer (&self->user, g_free);
g_clear_pointer (&self->password, g_free);
if (!self->use_direct_connection && self->state == CALLS_ACCOUNT_STATE_ONLINE)
go_online (CALLS_ACCOUNT (self), FALSE);
deinit_sip_account (self);
G_OBJECT_CLASS (calls_sip_origin_parent_class)->dispose (object);
}
static void
calls_sip_origin_finalize (GObject *object)
{
CallsSipOrigin *self = CALLS_SIP_ORIGIN (object);
g_hash_table_destroy (self->call_handles);
G_OBJECT_CLASS (calls_sip_origin_parent_class)->finalize (object);
}
static void
calls_sip_origin_class_init (CallsSipOriginClass *klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
object_class->constructed = calls_sip_origin_constructed;
object_class->dispose = calls_sip_origin_dispose;
object_class->finalize = calls_sip_origin_finalize;
object_class->get_property = calls_sip_origin_get_property;
object_class->set_property = calls_sip_origin_set_property;
props[PROP_ACC_HOST] =
g_param_spec_string ("host",
"Host",
"The host to connect to",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_HOST, props[PROP_ACC_HOST]);
props[PROP_ACC_USER] =
g_param_spec_string ("user",
"User",
"The username",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_USER, props[PROP_ACC_USER]);
props[PROP_ACC_PASSWORD] =
g_param_spec_string ("password",
"Password",
"The password",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_PASSWORD, props[PROP_ACC_PASSWORD]);
props[PROP_ACC_DISPLAY_NAME] =
g_param_spec_string ("display-name",
"Display name",
"The display name",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_DISPLAY_NAME, props[PROP_ACC_DISPLAY_NAME]);
props[PROP_ACC_PORT] =
g_param_spec_int ("port",
"Port",
"The port to connect to",
0, 65535, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_PORT, props[PROP_ACC_PORT]);
props[PROP_ACC_PROTOCOL] =
g_param_spec_string ("transport-protocol",
"Transport protocol",
"The transport protocol to use for the connection",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_PROTOCOL, props[PROP_ACC_PROTOCOL]);
props[PROP_ACC_AUTO_CONNECT] =
g_param_spec_boolean ("auto-connect",
"Auto connect",
"Whether to connect automatically",
TRUE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_AUTO_CONNECT, props[PROP_ACC_AUTO_CONNECT]);
props[PROP_ACC_DIRECT] =
g_param_spec_boolean ("direct-mode",
"Direct mode",
"Whether to use a direct connection (no SIP server)",
FALSE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_ACC_DIRECT, props[PROP_ACC_DIRECT]);
props[PROP_ACC_LOCAL_PORT] =
g_param_spec_int ("local-port",
"Local port",
"The local port to which the SIP stack binds to",
0, 65535, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY);
g_object_class_install_property (object_class, PROP_ACC_LOCAL_PORT, props[PROP_ACC_LOCAL_PORT]);
props[PROP_SIP_CONTEXT] =
g_param_spec_pointer ("sip-context",
"SIP context",
"The SIP context (sofia) used for our sip handles",
G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY);
g_object_class_install_property (object_class, PROP_SIP_CONTEXT, props[PROP_SIP_CONTEXT]);
props[PROP_CAN_TEL] =
g_param_spec_boolean ("can-tel",
"Can telephone",
"Whether to this account can be used for PSTN telephony",
FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_CAN_TEL, props[PROP_CAN_TEL]);
props[PROP_MEDIA_ENCRYPTION] =
g_param_spec_enum ("media-encryption",
"Media encryption",
"The media encryption mode",
SIP_TYPE_MEDIA_ENCRYPTION,
SIP_MEDIA_ENCRYPTION_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (object_class, PROP_MEDIA_ENCRYPTION, props[PROP_MEDIA_ENCRYPTION]);
#define IMPLEMENTS(ID, NAME) \
g_object_class_override_property (object_class, ID, NAME); \
props[ID] = g_object_class_find_property(object_class, NAME);
IMPLEMENTS (PROP_ID, "id");
IMPLEMENTS (PROP_NAME, "name");
IMPLEMENTS (PROP_CALLS, "calls");
IMPLEMENTS (PROP_COUNTRY_CODE, "country-code");
IMPLEMENTS (PROP_ACC_STATE, "account-state");
IMPLEMENTS (PROP_ACC_ADDRESS, "address");
#undef IMPLEMENTS
}
static void
calls_sip_origin_message_source_interface_init (CallsOriginInterface *iface)
{
}
static void
calls_sip_origin_origin_interface_init (CallsOriginInterface *iface)
{
iface->dial = dial;
iface->supports_protocol = supports_protocol;
}
static void
calls_sip_origin_accounts_interface_init (CallsAccountInterface *iface)
{
iface->go_online = go_online;
iface->get_address = get_address;
}
static void
calls_sip_origin_init (CallsSipOrigin *self)
{
const char *sip_test_env = g_getenv ("CALLS_SIP_TEST");
if (!sip_test_env || sip_test_env[0] == '\0') {
CallsNetworkWatch *nw = calls_network_watch_get_default ();
if (nw)
g_signal_connect_swapped (calls_network_watch_get_default (), "network-changed",
G_CALLBACK (recreate_sip), self);
else
g_warning ("Network watch unavailable. Unable to detect network changes.");
}
self->call_handles = g_hash_table_new (NULL, NULL);
}
void
calls_sip_origin_set_credentials (CallsSipOrigin *self,
const char *host,
const char *user,
const char *password,
const char *display_name,
const char *transport_protocol,
gint port,
SipMediaEncryption media_encryption,
gboolean can_tel,
gboolean auto_connect)
{
g_return_if_fail (CALLS_IS_SIP_ORIGIN (self));
if (self->direct_mode) {
g_warning ("Not allowed to update credentials when using direct mode");
return;
}
g_return_if_fail (host);
g_return_if_fail (user);
g_return_if_fail (password);
if (transport_protocol)
g_return_if_fail (protocol_is_valid (transport_protocol));
g_free (self->host);
self->host = g_strdup (host);
g_free (self->user);
self->user = g_strdup (user);
g_free (self->password);
self->password = g_strdup (password);
g_clear_pointer (&self->display_name, g_free);
if (display_name)
self->display_name = g_strdup (display_name);
g_free (self->transport_protocol);
if (transport_protocol)
self->transport_protocol = g_strdup (transport_protocol);
else
self->transport_protocol = g_strdup ("UDP");
self->port = port;
self->can_tel = can_tel;
self->media_encryption = media_encryption;
self->auto_connect = auto_connect;
update_name (self);
recreate_sip (self);
}
Reference in a new issue View git blame Copy permalink