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Guillaume Roguez authoredThis reverts commit 1e249e4f. The idea is "good" in theory, but the implementation of setSecure() causes an exception error. We need to find a better solution. Change-Id: I4062388782c07ba3d5a9ab8eb9df655be676d9e4
Guillaume Roguez authoredThis reverts commit 1e249e4f. The idea is "good" in theory, but the implementation of setSecure() causes an exception error. We need to find a better solution. Change-Id: I4062388782c07ba3d5a9ab8eb9df655be676d9e4
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sipvoiplink.cpp 46.66 KiB
/*
* Copyright (C) 2004-2016 Savoir-faire Linux Inc.
*
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Yun Liu <yun.liu@savoirfairelinux.com>
* Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
* Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
* Author: Guillaume Roguez <guillaume.roguez@savoirfairelinux.com>
*
* This program 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.
*
* This program 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 this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "sipvoiplink.h"
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "sdp.h"
#include "sipcall.h"
#include "sipaccount.h"
#include "ringdht/ringaccount.h"
#include "manager.h"
#if HAVE_SDES
#include "sdes_negotiator.h"
#endif
#include "im/instant_messaging.h"
#include "system_codec_container.h"
#include "audio/audio_rtp_session.h"
#ifdef RING_VIDEO
#include "video/video_rtp_session.h"
#include "client/videomanager.h"
#endif
#include "pres_sub_server.h"
#include "array_size.h"
#include "ip_utils.h"
#include "sip_utils.h"
#include "string_utils.h"
#include "logger.h"
#include <pjsip/sip_endpoint.h>
#include <pjsip/sip_uri.h>
#include <pjsip-simple/presence.h>
#include <pjsip-simple/publish.h>
#include <istream>
#include <algorithm>
namespace ring {
using sip_utils::CONST_PJ_STR;
/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
static pjsip_endpoint *endpt_;
static pjsip_module mod_ua_;
static void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status);
static void sdp_request_offer_cb(pjsip_inv_session *inv, const pjmedia_sdp_session *offer);
static void sdp_create_offer_cb(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer);
static void invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *e);
static void outgoing_request_forked_cb(pjsip_inv_session *inv, pjsip_event *e);
static void transaction_state_changed_cb(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
static std::shared_ptr<SIPCall> getCallFromInvite(pjsip_inv_session* inv);
decltype(getGlobalInstance<SIPVoIPLink>)& getSIPVoIPLink = getGlobalInstance<SIPVoIPLink>;
static void
handleIncomingOptions(pjsip_rx_data *rdata)
{
pjsip_tx_data *tdata;
if (pjsip_endpt_create_response(endpt_, rdata, PJSIP_SC_OK, NULL, &tdata) != PJ_SUCCESS)
return;
#define ADD_HDR(hdr) do { \
const pjsip_hdr *cap_hdr = hdr; \
if (cap_hdr) \
pjsip_msg_add_hdr (tdata->msg, (pjsip_hdr*) pjsip_hdr_clone (tdata->pool, cap_hdr)); \
} while (0)
#define ADD_CAP(cap) ADD_HDR(pjsip_endpt_get_capability(endpt_, cap, NULL));
ADD_CAP(PJSIP_H_ALLOW);
ADD_CAP(PJSIP_H_ACCEPT);
ADD_CAP(PJSIP_H_SUPPORTED);
ADD_HDR(pjsip_evsub_get_allow_events_hdr(NULL));
pjsip_response_addr res_addr;
pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
if (pjsip_endpt_send_response(endpt_, &res_addr, tdata, NULL, NULL) != PJ_SUCCESS)
pjsip_tx_data_dec_ref(tdata);
}
// return PJ_FALSE so that eventuall other modules will handle these requests
// TODO: move Voicemail to separate module
static pj_bool_t
transaction_response_cb(pjsip_rx_data *rdata)
{
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
if (!dlg)
return PJ_FALSE;
pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
if (!tsx or tsx->method.id != PJSIP_INVITE_METHOD)
return PJ_FALSE;
if (tsx->status_code / 100 == 2) {
/**
* Send an ACK message inside a transaction. PJSIP send automatically, non-2xx ACK response.
* ACK for a 2xx response must be send using this method.
*/
pjsip_tx_data *tdata;
if (rdata->msg_info.cseq) {
pjsip_dlg_create_request(dlg, &pjsip_ack_method, rdata->msg_info.cseq->cseq, &tdata);
pjsip_dlg_send_request(dlg, tdata, -1, NULL);
}
}
return PJ_FALSE;
}
static pj_status_t
try_respond_stateless(pjsip_endpoint *endpt, pjsip_rx_data *rdata, int st_code,
const pj_str_t *st_text, const pjsip_hdr *hdr_list,
const pjsip_msg_body *body)
{
/* Check that no UAS transaction has been created for this request.
* If UAS transaction has been created for this request, application
* MUST send the response statefully using that transaction.
*/
if (!pjsip_rdata_get_tsx(rdata))
return pjsip_endpt_respond_stateless(endpt, rdata, st_code, st_text, hdr_list, body);
else
RING_ERR("Transaction has been created for this request, send response "
"statefully instead");
return !PJ_SUCCESS;
}
static pj_bool_t
transaction_request_cb(pjsip_rx_data *rdata)
{
if (!rdata or !rdata->msg_info.msg) {
RING_ERR("rx_data is NULL");
return PJ_FALSE;
}
pjsip_method *method = &rdata->msg_info.msg->line.req.method;
if (!method) {
RING_ERR("method is NULL");
return PJ_FALSE;
}
if (method->id == PJSIP_ACK_METHOD && pjsip_rdata_get_dlg(rdata))
return PJ_FALSE;
if (!rdata->msg_info.to or !rdata->msg_info.from or !rdata->msg_info.via) {
RING_ERR("Missing From, To or Via fields");
return PJ_FALSE;
}
const auto sip_to_uri = reinterpret_cast<pjsip_sip_uri*>(pjsip_uri_get_uri(rdata->msg_info.to->uri));
const auto sip_from_uri = reinterpret_cast<pjsip_sip_uri*>(pjsip_uri_get_uri(rdata->msg_info.from->uri));
const pjsip_host_port& sip_via = rdata->msg_info.via->sent_by;
if (!sip_to_uri or !sip_from_uri or !sip_via.host.ptr) {
RING_ERR("NULL uri");
return PJ_FALSE;
}
std::string toUsername(sip_to_uri->user.ptr, sip_to_uri->user.slen);
std::string toHost(sip_to_uri->host.ptr, sip_to_uri->host.slen);
std::string viaHostname(sip_via.host.ptr, sip_via.host.slen);
const std::string remote_user(sip_from_uri->user.ptr, sip_from_uri->user.slen);
const std::string remote_hostname(sip_from_uri->host.ptr, sip_from_uri->host.slen);
char tmp[PJSIP_MAX_URL_SIZE];
size_t length = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, sip_from_uri, tmp, PJSIP_MAX_URL_SIZE);
std::string peerNumber(tmp, length);
sip_utils::stripSipUriPrefix(peerNumber);
if (not remote_user.empty() and not remote_hostname.empty())
peerNumber = remote_user + "@" + remote_hostname;
auto link = getSIPVoIPLink();
if (not link) {
RING_ERR("no more VoIP link");
return PJ_FALSE; }
auto account(link->guessAccount(toUsername, viaHostname, remote_hostname));
if (!account) {
RING_ERR("NULL account");
return PJ_FALSE;
}
const auto& account_id = account->getAccountID();
pjsip_msg_body *body = rdata->msg_info.msg->body;
if (method->id == PJSIP_OTHER_METHOD) {
pj_str_t *str = &method->name;
std::string request(str->ptr, str->slen);
if (request.find("NOTIFY") != std::string::npos) {
if (body and body->data) {
int voicemail = 0;
int ret = sscanf((const char*) body->data, "Voice-Message: %d/", &voicemail);
if (ret == 1 and voicemail != 0)
Manager::instance().startVoiceMessageNotification(account_id, voicemail);
}
} else if (request.find("MESSAGE") != std::string::npos) {
// Reply 200 immediatly (RFC 3428, ch. 7)
try_respond_stateless(endpt_, rdata, PJSIP_SC_OK, nullptr, nullptr, nullptr);
// Process message content in case of multi-part body
auto payloads = im::parseSipMessage(rdata->msg_info.msg);
if (payloads.size() > 0)
account->onTextMessage(peerNumber, payloads);
return PJ_FALSE;
}
try_respond_stateless(endpt_, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
return PJ_FALSE;
} else if (method->id == PJSIP_OPTIONS_METHOD) {
handleIncomingOptions(rdata);
return PJ_FALSE;
} else if (method->id != PJSIP_INVITE_METHOD && method->id != PJSIP_ACK_METHOD) {
try_respond_stateless(endpt_, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return PJ_FALSE;
}
pjmedia_sdp_session *r_sdp;
if (!body || pjmedia_sdp_parse(rdata->tp_info.pool, (char*) body->data, body->len, &r_sdp) != PJ_SUCCESS)
r_sdp = NULL;
if (not account->hasActiveCodec(MEDIA_AUDIO)) {
try_respond_stateless(endpt_, rdata, PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL, NULL);
return PJ_FALSE;
}
// Verify that we can handle the request
unsigned options = 0;
if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, endpt_, NULL) != PJ_SUCCESS) {
try_respond_stateless(endpt_, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return PJ_FALSE;
}
Manager::instance().hookPreference.runHook(rdata->msg_info.msg);
auto call = account->newIncomingCall(remote_user);
if (!call) {
return PJ_FALSE;
}
// RING_DBG("transaction_request_cb viaHostname %s toUsername %s addrToUse %s addrSdp %s peerNumber: %s" ,
// viaHostname.c_str(), toUsername.c_str(), addrToUse.toString().c_str(), addrSdp.toString().c_str(), peerNumber.c_str());
// Append PJSIP transport to the broker's SipTransport list
auto transport = link->sipTransportBroker->addTransport(rdata->tp_info.transport);
if (!transport) {
if (account->getAccountType() == SIPAccount::ACCOUNT_TYPE) {
RING_WARN("Using transport from account.");
transport = std::static_pointer_cast<SIPAccount>(account)->getTransport();
}
if (!transport) {
RING_ERR("No suitable transport to answer this call.");
return PJ_FALSE;
}
}
call->setTransport(transport);
// FIXME : for now, use the same address family as the SIP transport
auto family = pjsip_transport_type_get_af(pjsip_transport_get_type_from_flag(transport->get()->flag));
IpAddr addrToUse = ip_utils::getInterfaceAddr(account->getLocalInterface(), family);
IpAddr addrSdp;
if (account->getUPnPActive()) {
/* use UPnP addr, or published addr if its set */
addrSdp = account->getPublishedSameasLocal() ?
account->getUPnPIpAddress() : account->getPublishedIpAddress();
} else {
addrSdp = account->isStunEnabled() or (not account->getPublishedSameasLocal())
? account->getPublishedIpAddress() : addrToUse;
}
/* fallback on local address */
if (not addrSdp) addrSdp = addrToUse;
// Try to obtain display name from From: header first, fallback on Contact:
auto peerDisplayName = sip_utils::parseDisplayName(rdata->msg_info.from);
if (peerDisplayName.empty()) {
if (auto hdr = static_cast<const pjsip_contact_hdr*>(pjsip_msg_find_hdr(rdata->msg_info.msg,
PJSIP_H_CONTACT,
nullptr))) {
peerDisplayName = sip_utils::parseDisplayName(hdr);
}
}
call->setState(Call::ConnectionState::PROGRESSING);
call->setPeerNumber(peerNumber);
call->setPeerDisplayName(peerDisplayName);
call->initRecFilename(peerNumber);
call->setCallMediaLocal(addrToUse);
call->getSDP().setPublishedIP(addrSdp);
if (account->isStunEnabled())
call->updateSDPFromSTUN();
call->getSDP().receiveOffer(r_sdp,
account->getActiveAccountCodecInfoList(MEDIA_AUDIO),
account->getActiveAccountCodecInfoList(account->isVideoEnabled() ? MEDIA_VIDEO : MEDIA_NONE),
account->getSrtpKeyExchange()
);
auto ice_attrs = Sdp::getIceAttributes(r_sdp);
if (not ice_attrs.ufrag.empty() and not ice_attrs.pwd.empty()) {
if (not call->getIceTransport()) {
RING_DBG("Initializing ICE transport");
call->initIceTransport(false);
}
call->setupLocalSDPFromIce();
}
pjsip_dialog *dialog = nullptr;
if (pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, nullptr, &dialog) != PJ_SUCCESS) { RING_ERR("Could not create uas");
call.reset();
try_respond_stateless(endpt_, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, nullptr, nullptr, nullptr);
return PJ_FALSE;
}
pjsip_tpselector tp_sel = SIPVoIPLink::getTransportSelector(transport->get());
if (!dialog or pjsip_dlg_set_transport(dialog, &tp_sel) != PJ_SUCCESS) {
RING_ERR("Could not set transport for dialog");
return PJ_FALSE;
}
pjsip_inv_session* inv = nullptr;
pjsip_inv_create_uas(dialog, rdata, call->getSDP().getLocalSdpSession(), PJSIP_INV_SUPPORT_ICE, &inv);
if (!inv) {
RING_ERR("Call invite is not initialized");
return PJ_FALSE;
}
pjsip_dlg_inc_lock(inv->dlg);
inv->mod_data[mod_ua_.id] = call.get();
call->inv.reset(inv);
// Check whether Replaces header is present in the request and process accordingly.
pjsip_dialog *replaced_dlg;
pjsip_tx_data *response;
if (pjsip_replaces_verify_request(rdata, &replaced_dlg, PJ_FALSE, &response) != PJ_SUCCESS) {
RING_ERR("Something wrong with Replaces request.");
call.reset();
// Something wrong with the Replaces header.
if (response) {
pjsip_response_addr res_addr;
pjsip_get_response_addr(response->pool, rdata, &res_addr);
pjsip_endpt_send_response(endpt_, &res_addr, response,
NULL, NULL);
} else {
try_respond_stateless(endpt_, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
}
return PJ_FALSE;
}
// Check if call has been transfered
pjsip_tx_data *tdata = 0;
if (pjsip_inv_initial_answer(call->inv.get(), rdata, PJSIP_SC_TRYING, NULL, NULL, &tdata) != PJ_SUCCESS) {
RING_ERR("Could not create answer TRYING");
return PJ_FALSE;
}
if (pjsip_inv_send_msg(call->inv.get(), tdata) != PJ_SUCCESS) {
RING_ERR("Could not send msg TRYING");
return PJ_FALSE;
}
call->setState(Call::ConnectionState::TRYING);
if (pjsip_inv_answer(call->inv.get(), PJSIP_SC_RINGING, NULL, NULL, &tdata) != PJ_SUCCESS) {
RING_ERR("Could not create answer RINGING");
return PJ_FALSE;
}
// contactStr must stay in scope as long as tdata
const pj_str_t contactStr(account->getContactHeader(transport->get()));
sip_utils::addContactHeader(&contactStr, tdata);
if (pjsip_inv_send_msg(call->inv.get(), tdata) != PJ_SUCCESS) { RING_ERR("Could not send msg RINGING");
return PJ_FALSE;
}
call->setState(Call::ConnectionState::RINGING);
Manager::instance().incomingCall(*call, account_id);
if (replaced_dlg) {
// Get the INVITE session associated with the replaced dialog.
auto replaced_inv = pjsip_dlg_get_inv_session(replaced_dlg);
// Disconnect the "replaced" INVITE session.
if (pjsip_inv_end_session(replaced_inv, PJSIP_SC_GONE, nullptr, &tdata) == PJ_SUCCESS && tdata) {
pjsip_inv_send_msg(replaced_inv, tdata);
}
// Close call at application level
if (auto replacedCall = getCallFromInvite(replaced_inv))
replacedCall->hangup(PJSIP_SC_OK);
}
return PJ_FALSE;
}
static void
tp_state_callback(pjsip_transport* tp, pjsip_transport_state state,
const pjsip_transport_state_info* info)
{
// There is no way (at writing) to link a user data to a PJSIP transport.
// So we obtain it from the global SIPVoIPLink instance that owns it.
// Be sure the broker's owner is not deleted during proccess
if (auto sipLink = getSIPVoIPLink()) {
if (auto& broker = sipLink->sipTransportBroker)
broker->transportStateChanged(tp, state, info);
else
RING_ERR("SIPVoIPLink with invalid SipTransportBroker");
} else
RING_ERR("no more VoIP link");
}
/*************************************************************************************************/
pjsip_endpoint * SIPVoIPLink::getEndpoint()
{
return endpt_;
}
pjsip_module * SIPVoIPLink::getMod()
{
return &mod_ua_;
}
pj_pool_t*
SIPVoIPLink::getPool() noexcept
{
return pool_.get();
}
pj_caching_pool*
SIPVoIPLink::getCachingPool() noexcept
{
return &cp_;
}
SIPVoIPLink::SIPVoIPLink() : pool_(nullptr, pj_pool_release)
{
#define TRY(ret) do { \
if (ret != PJ_SUCCESS) \ throw VoipLinkException(#ret " failed"); \
} while (0)
pj_caching_pool_init(&cp_, &pj_pool_factory_default_policy, 0);
pool_.reset(pj_pool_create(&cp_.factory, PACKAGE, 4096, 4096, nullptr));
if (!pool_)
throw VoipLinkException("UserAgent: Could not initialize memory pool");
TRY(pjsip_endpt_create(&cp_.factory, pj_gethostname()->ptr, &endpt_));
auto ns = ip_utils::getLocalNameservers();
if (not ns.empty()) {
std::vector<pj_str_t> dns_nameservers(ns.size());
for (unsigned i=0, n=ns.size(); i<n; i++) {
char hbuf[NI_MAXHOST];
getnameinfo((sockaddr*)&ns[i], ns[i].getLength(), hbuf, sizeof(hbuf), nullptr, 0, NI_NUMERICHOST);
RING_DBG("Using SIP nameserver: %s", hbuf);
pj_strdup2(pool_.get(), &dns_nameservers[i], hbuf);
}
pj_dns_resolver* resv;
TRY(pjsip_endpt_create_resolver(endpt_, &resv));
TRY(pj_dns_resolver_set_ns(resv, ns.size(), dns_nameservers.data(), nullptr));
TRY(pjsip_endpt_set_resolver(endpt_, resv));
}
sipTransportBroker.reset(new SipTransportBroker(endpt_, cp_, *pool_));
auto status = pjsip_tpmgr_set_state_cb(pjsip_endpt_get_tpmgr(endpt_),
tp_state_callback);
if (status != PJ_SUCCESS)
RING_ERR("Can't set transport callback: %s", sip_utils::sip_strerror(status).c_str());
if (!ip_utils::getLocalAddr())
throw VoipLinkException("UserAgent: Unable to determine network capabilities");
TRY(pjsip_tsx_layer_init_module(endpt_));
TRY(pjsip_ua_init_module(endpt_, nullptr));
TRY(pjsip_replaces_init_module(endpt_)); // See the Replaces specification in RFC 3891
TRY(pjsip_100rel_init_module(endpt_));
// Initialize and register ring module
mod_ua_.name = pj_str((char*) PACKAGE);
mod_ua_.id = -1;
mod_ua_.priority = PJSIP_MOD_PRIORITY_APPLICATION;
mod_ua_.on_rx_request = &transaction_request_cb;
mod_ua_.on_rx_response = &transaction_response_cb;
TRY(pjsip_endpt_register_module(endpt_, &mod_ua_));
TRY(pjsip_evsub_init_module(endpt_));
TRY(pjsip_xfer_init_module(endpt_));
// presence/publish management
TRY(pjsip_pres_init_module(endpt_, pjsip_evsub_instance()));
TRY(pjsip_endpt_register_module(endpt_, &PresSubServer::mod_presence_server));
static const pjsip_inv_callback inv_cb = {
invite_session_state_changed_cb,
outgoing_request_forked_cb,
transaction_state_changed_cb,
sdp_request_offer_cb,
#if PJ_VERSION_NUM > (2 << 24 | 1 << 16)
nullptr /* on_rx_reinvite */,
#endif
sdp_create_offer_cb,
sdp_media_update_cb,
nullptr /* on_send_ack */,
nullptr /* on_redirected */,
};
TRY(pjsip_inv_usage_init(endpt_, &inv_cb));
static const pj_str_t allowed[] = {
CONST_PJ_STR("INFO"),
CONST_PJ_STR("OPTIONS"),
CONST_PJ_STR("MESSAGE"),
CONST_PJ_STR("PUBLISH"),
};
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ALLOW, nullptr, PJ_ARRAY_SIZE(allowed), allowed);
static const pj_str_t text_plain = CONST_PJ_STR("text/plain");
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, nullptr, 1, &text_plain);
static const pj_str_t accepted = CONST_PJ_STR("application/sdp");
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, nullptr, 1, &accepted);
static const pj_str_t iscomposing = CONST_PJ_STR("application/im-iscomposing+xml");
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, nullptr, 1, &iscomposing);
TRY(pjsip_replaces_init_module(endpt_));
#undef TRY
// ready to handle events
// Implementation note: we don't use std::bind(xxx, this) here
// as handleEvents needs a valid instance to be called.
Manager::instance().registerEventHandler((uintptr_t)this,
[this]{ handleEvents(); });
RING_DBG("SIPVoIPLink@%p", this);
}
SIPVoIPLink::~SIPVoIPLink()
{
RING_DBG("~SIPVoIPLink@%p", this);
// Remaining calls should not happen as possible upper callbacks
// may be called and another instance of SIPVoIPLink can be re-created!
if (not Manager::instance().callFactory.empty<SIPCall>())
RING_ERR("%zu SIP calls remains!",
Manager::instance().callFactory.callCount<SIPCall>());
sipTransportBroker->shutdown();
const int MAX_TIMEOUT_ON_LEAVING = 5;
for (int timeout = 0;
pjsip_tsx_layer_get_tsx_count() and timeout < MAX_TIMEOUT_ON_LEAVING;
timeout++)
sleep(1);
pjsip_tpmgr_set_state_cb(pjsip_endpt_get_tpmgr(endpt_), nullptr);
Manager::instance().unregisterEventHandler((uintptr_t)this);
handleEvents();
sipTransportBroker.reset();
pjsip_endpt_destroy(endpt_);
pool_.reset();
pj_caching_pool_destroy(&cp_);
RING_DBG("destroying SIPVoIPLink@%p", this);
}
std::shared_ptr<SIPAccountBase>
SIPVoIPLink::guessAccount(const std::string& userName,
const std::string& server,
const std::string& fromUri) const
{
RING_DBG("username = %s, server = %s, from = %s", userName.c_str(), server.c_str(), fromUri.c_str());
// Try to find the account id from username and server name by full match
std::shared_ptr<SIPAccountBase> result;
std::shared_ptr<SIPAccountBase> IP2IPAccount;
MatchRank best = MatchRank::NONE;
// DHT accounts
for (const auto& account : Manager::instance().getAllAccounts<RingAccount>()) {
if (!account)
continue;
const MatchRank match(account->matches(userName, server));
// return right away if this is a full match
if (match == MatchRank::FULL) {
return account;
} else if (match > best) {
best = match;
result = account;
}
}
// SIP accounts
for (const auto& account : Manager::instance().getAllAccounts<SIPAccount>()) {
if (!account)
continue;
const MatchRank match(account->matches(userName, server, endpt_, pool_.get()));
// return right away if this is a full match
if (match == MatchRank::FULL) {
return account;
} else if (match > best) {
best = match;
result = account;
} else if (!IP2IPAccount && account->isIP2IP()) {
// Allow IP2IP calls if an account exists for this type of calls
IP2IPAccount = account;
}
}
return result ? result : IP2IPAccount;
}
// Called from EventThread::run (not main thread)
void
SIPVoIPLink::handleEvents()
{
// We have to register the external thread so it could access the pjsip frameworks
if (!pj_thread_is_registered()) {
#if __GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)
static thread_local pj_thread_desc desc;
static thread_local pj_thread_t *this_thread;
#else
static __thread pj_thread_desc desc;
static __thread pj_thread_t *this_thread;
#endif
RING_DBG("Registering thread");
pj_thread_register(NULL, desc, &this_thread);
}
static const pj_time_val timeout = {0, 0}; // polling
auto ret = pjsip_endpt_handle_events(endpt_, &timeout);
if (ret != PJ_SUCCESS)
RING_ERR("pjsip_endpt_handle_events failed with error %s",
sip_utils::sip_strerror(ret).c_str());
#ifdef RING_VIDEO
dequeKeyframeRequests();
#endif
}
void SIPVoIPLink::registerKeepAliveTimer(pj_timer_entry &timer, pj_time_val &delay)
{ RING_DBG("Register new keep alive timer %d with delay %ld", timer.id, delay.sec);
if (timer.id == -1)
RING_WARN("Timer already scheduled");
switch (pjsip_endpt_schedule_timer(endpt_, &timer, &delay)) {
case PJ_SUCCESS:
break;
default:
RING_ERR("Could not schedule new timer in pjsip endpoint");
/* fallthrough */
case PJ_EINVAL:
RING_ERR("Invalid timer or delay entry");
break;
case PJ_EINVALIDOP:
RING_ERR("Invalid timer entry, maybe already scheduled");
break;
}
}
void SIPVoIPLink::cancelKeepAliveTimer(pj_timer_entry& timer)
{
pjsip_endpt_cancel_timer(endpt_, &timer);
}
#ifdef RING_VIDEO
// Called from a video thread
void
SIPVoIPLink::enqueueKeyframeRequest(const std::string &id)
{
if (auto link = getSIPVoIPLink()) {
std::lock_guard<std::mutex> lock(link->keyframeRequestsMutex_);
link->keyframeRequests_.push(id);
} else
RING_ERR("no more VoIP link");
}
// Called from SIP event thread
void
SIPVoIPLink::dequeKeyframeRequests()
{
int max_requests = 20;
while (not keyframeRequests_.empty() and max_requests--) {
std::lock_guard<std::mutex> lock(keyframeRequestsMutex_);
const std::string &id(keyframeRequests_.front());
requestKeyframe(id);
keyframeRequests_.pop();
}
}
// Called from SIP event thread
void
SIPVoIPLink::requestKeyframe(const std::string &callID)
{
std::shared_ptr<SIPCall> call;
const int tries = 10;
for (int i = 0; !call and i < tries; ++i)
call = Manager::instance().callFactory.getCall<SIPCall>(callID); // fixme: need a try version
if (!call)
return;
const char * const BODY =
"<?xml version=\"1.0\" encoding=\"utf-8\" ?>"
"<media_control><vc_primitive><to_encoder>" "<picture_fast_update/>"
"</to_encoder></vc_primitive></media_control>";
RING_DBG("Sending video keyframe request via SIP INFO");
call->sendSIPInfo(BODY, "media_control+xml");
}
#endif
///////////////////////////////////////////////////////////////////////////////
// Private functions
///////////////////////////////////////////////////////////////////////////////
static std::shared_ptr<SIPCall>
getCallFromInvite(pjsip_inv_session* inv)
{
if (auto call_ptr = static_cast<SIPCall*>(inv->mod_data[mod_ua_.id]))
return std::static_pointer_cast<SIPCall>(call_ptr->shared_from_this());
return nullptr;
}
static void
invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *ev)
{
auto call = getCallFromInvite(inv);
if (not call)
return;
if (ev->type != PJSIP_EVENT_TSX_STATE and ev->type != PJSIP_EVENT_TX_MSG) {
RING_WARN("[call:%s] INVITE@%p state changed to %d (%s): unwaited event type %d",
call->getCallId().c_str(), inv, inv->state, pjsip_inv_state_name(inv->state),
ev->type);
return;
}
decltype(pjsip_transaction::status_code) status_code;
if (ev->type != PJSIP_EVENT_TX_MSG) {
const auto tsx = ev->body.tsx_state.tsx;
status_code = tsx ? tsx->status_code : PJSIP_SC_NOT_FOUND;
const pj_str_t* description = pjsip_get_status_text(status_code);
RING_DBG("[call:%s] INVITE@%p state changed to %d (%s): cause=%d, tsx@%p status %d (%.*s)",
call->getCallId().c_str(), inv, inv->state, pjsip_inv_state_name(inv->state),
inv->cause, tsx, status_code, (int)description->slen, description->ptr);
} else {
status_code = 0;
RING_DBG("[call:%s] INVITE@%p state changed to %d (%s): cause=%d (TX_MSG)",
call->getCallId().c_str(), inv, inv->state, pjsip_inv_state_name(inv->state),
inv->cause);
}
switch (inv->state) {
case PJSIP_INV_STATE_EARLY:
if (status_code == PJSIP_SC_RINGING)
call->onPeerRinging();
break;
case PJSIP_INV_STATE_CONFIRMED:
// After we sent or received a ACK - The connection is established
call->onAnswered();
break;
case PJSIP_INV_STATE_DISCONNECTED:
switch (inv->cause) {
// When the peer manually refuse the call
case PJSIP_SC_DECLINE:
case PJSIP_SC_BUSY_EVERYWHERE:
case PJSIP_SC_BUSY_HERE:
if (inv->role != PJSIP_ROLE_UAC)
break; // close call
// The call terminates normally - BYE / CANCEL
case PJSIP_SC_OK:
case PJSIP_SC_REQUEST_TERMINATED:
call->onClosed();
break;
// Error/unhandled conditions
default:
call->onFailure(inv->cause);
break;
}
break;
default:
break;
}
}
static void
sdp_request_offer_cb(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
{
if (auto call = getCallFromInvite(inv))
call->onReceiveOffer(offer);
}
static void
sdp_create_offer_cb(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
auto call = getCallFromInvite(inv);
if (not call)
return;
const auto& account = call->getSIPAccount();
auto family = pj_AF_INET();
// FIXME : for now, use the same address family as the SIP transport
if (auto dlg = inv->dlg) {
if (dlg->tp_sel.type == PJSIP_TPSELECTOR_TRANSPORT) {
if (auto tr = dlg->tp_sel.u.transport)
family = tr->local_addr.addr.sa_family;
} else if (dlg->tp_sel.type == PJSIP_TPSELECTOR_TRANSPORT) {
if (auto tr = dlg->tp_sel.u.listener)
family = tr->local_addr.addr.sa_family;
}
}
auto ifaceAddr = ip_utils::getInterfaceAddr(account.getLocalInterface(), family);
IpAddr address;
if (account.getUPnPActive()) {
/* use UPnP addr, or published addr if its set */
address = account.getPublishedSameasLocal() ?
account.getUPnPIpAddress() : account.getPublishedIpAddress();
} else {
address = account.getPublishedSameasLocal() ?
ifaceAddr : account.getPublishedIpAddress();
}
/* fallback on local address */
if (not address) address = ifaceAddr;
call->setCallMediaLocal(address);
auto& localSDP = call->getSDP();
localSDP.setPublishedIP(address);
const bool created = localSDP.createOffer(
account.getActiveAccountCodecInfoList(MEDIA_AUDIO),
account.getActiveAccountCodecInfoList(account.isVideoEnabled() ? MEDIA_VIDEO : MEDIA_NONE),
account.getSrtpKeyExchange()
);
if (created)
*p_offer = localSDP.getLocalSdpSession();
}
static void
dump_sdp_session(const pjmedia_sdp_session* sdp_session, const char* header)
{
char buffer[4096] {};
if (pjmedia_sdp_print(sdp_session, buffer, sizeof buffer) == -1) {
RING_ERR("%sSDP too big for dump", header);
return;
}
RING_DBG("%s%s", header, buffer);
}
static const pjmedia_sdp_session*
get_active_remote_sdp(pjsip_inv_session *inv)
{
const pjmedia_sdp_session* sdp_session {};
if (pjmedia_sdp_neg_get_active_remote(inv->neg, &sdp_session) != PJ_SUCCESS) {
RING_ERR("Active remote not present");
return nullptr;
}
if (pjmedia_sdp_validate(sdp_session) != PJ_SUCCESS) {
RING_ERR("Invalid remote SDP session");
return nullptr;
}
dump_sdp_session(sdp_session, "Remote active SDP Session:\n");
return sdp_session;
}
static const pjmedia_sdp_session*
get_active_local_sdp(pjsip_inv_session *inv)
{
const pjmedia_sdp_session* sdp_session {};
if (pjmedia_sdp_neg_get_active_local(inv->neg, &sdp_session) != PJ_SUCCESS) {
RING_ERR("Active local not present");
return nullptr;
}
if (pjmedia_sdp_validate(sdp_session) != PJ_SUCCESS) {
RING_ERR("Invalid local SDP session");
return nullptr;
}
dump_sdp_session(sdp_session, "Local active SDP Session:\n");
return sdp_session;
}
// This callback is called after SDP offer/answer session has completed.
static void
sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
{
auto call = getCallFromInvite(inv);
if (not call)
return;
RING_DBG("[call:%s] INVITE@%p media update: status %d",
call->getCallId().c_str(), inv, status);
if (status != PJ_SUCCESS) {
const int reason = inv->state != PJSIP_INV_STATE_NULL and
inv->state != PJSIP_INV_STATE_CONFIRMED ? PJSIP_SC_UNSUPPORTED_MEDIA_TYPE : 0;
RING_WARN("[call:%s] SDP offer failed, reason %d", call->getCallId().c_str(), reason);
call->hangup(reason);
return;
}
// Fetch SDP data from request
const auto localSDP = get_active_local_sdp(inv);
const auto remoteSDP = get_active_remote_sdp(inv);
// Update our SDP manager
auto& sdp = call->getSDP();
sdp.setActiveLocalSdpSession(localSDP);
sdp.setActiveRemoteSdpSession(remoteSDP);
call->onMediaUpdate();
}
static void
outgoing_request_forked_cb(pjsip_inv_session * /*inv*/, pjsip_event * /*e*/)
{}
static bool
handleMediaControl(SIPCall& call, pjsip_msg_body* body)
{
/*
* Incoming INFO request for media control.
*/
const pj_str_t STR_APPLICATION = CONST_PJ_STR("application");
const pj_str_t STR_MEDIA_CONTROL_XML = CONST_PJ_STR("media_control+xml");
if (body->len and pj_stricmp(&body->content_type.type, &STR_APPLICATION) == 0 and
pj_stricmp(&body->content_type.subtype, &STR_MEDIA_CONTROL_XML) == 0) {
pj_str_t control_st;
/* Apply and answer the INFO request */
pj_strset(&control_st, (char *) body->data, body->len);
const pj_str_t PICT_FAST_UPDATE = CONST_PJ_STR("picture_fast_update");
if (pj_strstr(&control_st, &PICT_FAST_UPDATE)) {
#ifdef RING_VIDEO
RING_DBG("handling picture fast update request");
call.getVideoRtp().forceKeyFrame();
#endif
return true;
}
}
return false;
}
/**
* Helper function to process refer function on call transfer
*/
static bool
transferCall(SIPCall& call, const std::string& refer_to)
{
const auto& callId = call.getCallId();
RING_WARN("[call:%s] Trying to transfer to %s", callId.c_str(), refer_to.c_str());
try {
Manager::instance().newOutgoingCall(refer_to, call.getAccountId());
Manager::instance().hangupCall(callId);
} catch (const std::exception& e) {
RING_ERR("[call:%s] SIP transfer failed: %s", callId.c_str(), e.what());
return false;
}
return true;
}
static void
replyToRequest(pjsip_inv_session* inv, pjsip_rx_data* rdata, int status_code)
{
const auto ret = pjsip_dlg_respond(inv->dlg, rdata, status_code, nullptr, nullptr, nullptr);
if (ret != PJ_SUCCESS)
RING_WARN("SIP: failed to reply %d to request", status_code);
}
static void
onRequestRefer(pjsip_inv_session* inv, pjsip_rx_data* rdata, pjsip_msg* msg, SIPCall& call)
{
static const pj_str_t str_refer_to = CONST_PJ_STR("Refer-To");
if (auto refer_to = static_cast<pjsip_generic_string_hdr*>(pjsip_msg_find_hdr_by_name(msg, &str_refer_to, nullptr))) {
// RFC 3515, 2.4.2: reply bad request if no or too many refer-to header.
if (static_cast<void*>(refer_to->next) == static_cast<void*>(&msg->hdr) or
!pjsip_msg_find_hdr_by_name(msg, &str_refer_to, refer_to->next)) {
replyToRequest(inv, rdata, PJSIP_SC_ACCEPTED);
transferCall(call, std::string(refer_to->hvalue.ptr, refer_to->hvalue.slen));
// RFC 3515, 2.4.4: we MUST handle the processing using NOTIFY msgs
// But your current design doesn't permit that
return;
} else
RING_ERR("[call:%s] REFER: too many Refer-To headers", call.getCallId().c_str());
} else
RING_ERR("[call:%s] REFER: no Refer-To header", call.getCallId().c_str());
replyToRequest(inv, rdata, PJSIP_SC_BAD_REQUEST);
}
static void
onRequestInfo(pjsip_inv_session* inv, pjsip_rx_data* rdata, pjsip_msg* msg, SIPCall& call)
{
if (!msg->body or handleMediaControl(call, msg->body))
replyToRequest(inv, rdata, PJSIP_SC_OK);
}
static void
onRequestNotify(pjsip_inv_session* /*inv*/, pjsip_rx_data* /*rdata*/, pjsip_msg* msg, SIPCall& call)
{
if (!msg->body)
return;
const std::string bodyText {static_cast<char *>(msg->body->data), msg->body->len};
RING_DBG("[call:%s] NOTIFY body start - %p\n%s\n[call:%s] NOTIFY body end - %p",
call.getCallId().c_str(), msg->body,
bodyText.c_str(),
call.getCallId().c_str(), msg->body);
// TODO
}
static void
onRequestMessage(pjsip_inv_session* /*inv*/, pjsip_rx_data* /*rdata*/, pjsip_msg* msg,
SIPCall& call)
{
if (!msg->body)
return;
//TODO: for now we assume that the "from" is the message sender, this may not be true in the
// case of conferences; a content type containing this info will be added to the messages
// in the future
Manager::instance().incomingMessage(call.getCallId(), call.getPeerNumber(),
im::parseSipMessage(msg));
}
static void
transaction_state_changed_cb(pjsip_inv_session* inv, pjsip_transaction* tsx, pjsip_event* event){
auto call = getCallFromInvite(inv);
if (not call)
return;
// We process here only incoming request message
if (tsx->role != PJSIP_ROLE_UAS
or tsx->state != PJSIP_TSX_STATE_TRYING
or event->body.tsx_state.type != PJSIP_EVENT_RX_MSG) {
RING_DBG("[INVITE:%p] tsx_role=%d, tsx_state=%d, ev_type=%d, tsx_state_type=%d", inv,
tsx->role, tsx->state, event->type, event->body.tsx_state.type);
return;
}
const auto rdata = event->body.tsx_state.src.rdata;
if (!rdata) {
RING_ERR("[INVITE:%p] SIP RX request without rx data", inv);
return;
}
const auto msg = rdata->msg_info.msg;
if (msg->type != PJSIP_REQUEST_MSG) {
RING_ERR("[INVITE:%p] SIP RX request without msg", inv);
return;
}
// Using method name to dispatch
const std::string methodName {msg->line.req.method.name.ptr, (unsigned)msg->line.req.method.name.slen};
RING_DBG("[INVITE:%p] RX SIP method %d (%s)", inv, msg->line.req.method.id, methodName.c_str());
#ifdef DEBUG_SIP_REQUEST_MSG
char msgbuf[1000];
pjsip_msg_print(msg, msgbuf, sizeof msgbuf);
RING_DBG("%s", msgbuf);
#endif // DEBUG_SIP_MESSAGE
if (methodName == "REFER")
onRequestRefer(inv, rdata, msg, *call);
else if (methodName == "INFO")
onRequestInfo(inv, rdata, msg, *call);
else if (methodName == "NOTIFY")
onRequestNotify(inv, rdata, msg, *call);
else if (methodName == "MESSAGE")
onRequestMessage(inv, rdata, msg, *call);
}
int SIPVoIPLink::getModId()
{
return mod_ua_.id;
}
void SIPVoIPLink::createSDPOffer(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
assert(inv and p_offer);
sdp_create_offer_cb(inv, p_offer);
}
// Thread-safe DNS resolver callback mapping
class SafeResolveCallbackMap {
public:
using ResolveCallback = std::function<void(pj_status_t, const pjsip_server_addresses*)>;
void registerCallback(uintptr_t key, ResolveCallback&& cb) {
std::lock_guard<std::mutex> lk(mutex_);
cbMap_.emplace(key, std::move(cb));
}
void process(uintptr_t key, pj_status_t status, const pjsip_server_addresses* addr) {
std::lock_guard<std::mutex> lk(mutex_);
auto it = cbMap_.find(key); if (it != cbMap_.end()) {
it->second(status, addr);
cbMap_.erase(it);
}
}
private:
std::mutex mutex_;
std::map<uintptr_t, ResolveCallback> cbMap_;
};
static SafeResolveCallbackMap&
getResolveCallbackMap()
{
static SafeResolveCallbackMap map;
return map;
}
static void
resolver_callback(pj_status_t status, void *token, const struct pjsip_server_addresses* addr)
{
getResolveCallbackMap().process((uintptr_t)token, status, addr);
}
void
SIPVoIPLink::resolveSrvName(const std::string &name, pjsip_transport_type_e type, SrvResolveCallback cb)
{
if (name.length() >= PJ_MAX_HOSTNAME) {
RING_ERR("Hostname is too long");
cb({});
return;
}
// extract port if name is in form "server:port"
int port;
pj_ssize_t name_size;
const auto n = name.rfind(':');
if (n != std::string::npos) {
port = std::atoi(name.c_str() + n + 1);
name_size = n;
} else {
port = 0;
name_size = name.size();
}
RING_DBG("try to resolve '%s' (port: %u)", name.c_str(), port);
pjsip_host_info host_info {
.flag = 0,
.type = type,
.addr = {{(char*)name.c_str(), name_size}, port},
};
const auto token = std::hash<std::string>()(name + to_string(type));
getResolveCallbackMap().registerCallback(token,
[=](pj_status_t s, const pjsip_server_addresses* r) {
try {
if (s != PJ_SUCCESS || !r) {
RING_WARN("Can't resolve \"%s\" using pjsip_endpt_resolve, trying getaddrinfo.", name.c_str());
std::thread([=](){
auto ips = ip_utils::getAddrList(name.c_str());
runOnMainThread(std::bind(cb, ips.empty() ? std::vector<IpAddr>{} : std::move(ips)));
}).detach();
} else {
std::vector<IpAddr> ips;
ips.reserve(r->count);
for (unsigned i=0; i < r->count; i++)
ips.push_back(r->entry[i].addr);
cb(ips);
}
} catch (const std::exception& e) { RING_ERR("Error resolving address: %s", e.what());
cb({});
}
});
pjsip_endpt_resolve(endpt_, pool_.get(), &host_info, (void*)token, resolver_callback);
}
#define RETURN_IF_NULL(A, M, ...) \
if ((A) == NULL) { RING_WARN(M, ##__VA_ARGS__); return; }
#define RETURN_FALSE_IF_NULL(A, M, ...) \
if ((A) == NULL) { RING_WARN(M, ##__VA_ARGS__); return false; }
void
SIPVoIPLink::findLocalAddressFromTransport(pjsip_transport* transport,
pjsip_transport_type_e transportType,
const std::string& host,
std::string& addr,
pj_uint16_t& port) const
{
// Initialize the sip port with the default SIP port
port = pjsip_transport_get_default_port_for_type(transportType);
// Initialize the sip address with the hostname
const auto pjMachineName = pj_gethostname();
addr = std::string(pjMachineName->ptr, pjMachineName->slen);
// Update address and port with active transport
RETURN_IF_NULL(transport,
"Transport is NULL in findLocalAddress, using local address %s :%d",
addr.c_str(), port);
// get the transport manager associated with the SIP enpoint
auto tpmgr = pjsip_endpt_get_tpmgr(endpt_);
RETURN_IF_NULL(tpmgr,
"Transport manager is NULL in findLocalAddress, using local address %s :%d",
addr.c_str(), port);
pj_str_t pjstring;
pj_cstr(&pjstring, host.c_str());
auto tp_sel = getTransportSelector(transport);
pjsip_tpmgr_fla2_param param = { transportType, &tp_sel, pjstring, PJ_FALSE,
{nullptr, 0}, 0, nullptr };
if (pjsip_tpmgr_find_local_addr2(tpmgr, pool_.get(), ¶m) != PJ_SUCCESS) {
RING_WARN("Could not retrieve local address and port from transport, using %s :%d",
addr.c_str(), port);
return;
}
// Update local address based on the transport type
addr = std::string(param.ret_addr.ptr, param.ret_addr.slen);
// Determine the local port based on transport information
port = param.ret_port;
}
bool
SIPVoIPLink::findLocalAddressFromSTUN(pjsip_transport* transport,
pj_str_t* stunServerName,
int stunPort,
std::string& addr,
pj_uint16_t& port) const
{
// WARN: this code use pjstun_get_mapped_addr2 that works
// in IPv4 only.
// WARN: this function is blocking (network request).
// Initialize the sip port with the default SIP port port = sip_utils::DEFAULT_SIP_PORT;
// Get Local IP address
auto localIp = ip_utils::getLocalAddr(pj_AF_INET());
if (not localIp) {
RING_WARN("Failed to find local IP");
return false;
}
addr = localIp.toString();
// Update address and port with active transport
RETURN_FALSE_IF_NULL(transport,
"Transport is NULL in findLocalAddress, using local address %s:%u",
addr.c_str(), port);
RING_DBG("STUN mapping of '%s:%u'", addr.c_str(), port);
pj_sockaddr_in mapped_addr;
pj_sock_t sipSocket = pjsip_udp_transport_get_socket(transport);
const pjstun_setting stunOpt = {PJ_TRUE, *stunServerName, stunPort,
*stunServerName, stunPort};
const pj_status_t stunStatus = pjstun_get_mapped_addr2(&cp_.factory,
&stunOpt, 1,
&sipSocket,
&mapped_addr);
switch (stunStatus) {
case PJLIB_UTIL_ESTUNNOTRESPOND:
RING_ERR("No response from STUN server %.*s",
(int)stunServerName->slen, stunServerName->ptr);
return false;
case PJLIB_UTIL_ESTUNSYMMETRIC:
RING_ERR("Different mapped addresses are returned by servers.");
return false;
case PJ_SUCCESS:
port = pj_sockaddr_in_get_port(&mapped_addr);
addr = IpAddr((const pj_sockaddr&)mapped_addr).toString();
RING_DBG("STUN server %.*s replied '%s:%u'",
(int)stunServerName->slen, stunServerName->ptr,
addr.c_str(), port);
return true;
default: // use given address, silent any not handled error
RING_WARN("Error from STUN server %.*s, using source address",
(int)stunServerName->slen, stunServerName->ptr);
return false;
}
}
#undef RETURN_IF_NULL
#undef RETURN_FALSE_IF_NULL
} // namespace ring