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f9c997d28dd03616f59e54b74dc078b0953d8f98
jami-daemon/daemon/src/sip/sipvoiplink.cpp
Tristan Matthews f9c997d28d * #12475: videortp: create videortpsession on the stack
2012-06-21 11:16:45 -04:00

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/*
* Copyright (C) 2004, 2005, 2006, 2008, 2009, 2010 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>
*
* 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., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* Additional permission under GNU GPL version 3 section 7:
*
* If you modify this program, or any covered work, by linking or
* combining it with the OpenSSL project's OpenSSL library (or a
* modified version of that library), containing parts covered by the
* terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
* grants you additional permission to convey the resulting work.
* Corresponding Source for a non-source form of such a combination
* shall include the source code for the parts of OpenSSL used as well
* as that of the covered work.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "sip_utils.h"
#include "sipvoiplink.h"
#include "array_size.h"
#include "manager.h"
#include "logger.h"
#include "sip/sdp.h"
#include "sipcall.h"
#include "sipaccount.h"
#include "eventthread.h"
#include "sdes_negotiator.h"
#include "array_size.h"
#include "dbus/dbusmanager.h"
#include "dbus/callmanager.h"
#include "dbus/configurationmanager.h"
#include "im/instant_messaging.h"
#include "audio/audiolayer.h"
#ifdef SFL_VIDEO
#include "video/video_rtp_session.h"
#endif
#include "pjsip/sip_endpoint.h"
#include "pjsip/sip_transport_tls.h"
#include "pjsip/sip_uri.h"
#include "pjnath.h"
#include <netinet/in.h>
#include <arpa/nameser.h>
#include <arpa/inet.h>
#include <resolv.h>
#include <istream>
#include <utility> // for std::pair
#include <map>
using namespace sfl;
SIPVoIPLink *SIPVoIPLink::instance_ = 0;
bool SIPVoIPLink::destroyed_ = false;
namespace {
/** A map to retreive SFLphone internal call id
* Given a SIP call ID (usefull for transaction sucha as transfer)*/
static std::map<std::string, std::string> transferCallID;
/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
/**
* Set audio and video (SDP) configuration for a call
* localport, localip, localexternalport
* @param call a SIPCall valid pointer
*/
void setCallMediaLocal(SIPCall* call, const std::string &localIP);
static pj_caching_pool pool_cache, *cp_ = &pool_cache;
static pj_pool_t *pool_;
static pjsip_endpoint *endpt_;
static pjsip_module mod_ua_;
static pj_thread_t *thread_;
void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status);
void sdp_request_offer_cb(pjsip_inv_session *inv, const pjmedia_sdp_session *offer);
void sdp_create_offer_cb(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer);
void invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *e);
void outgoing_request_forked_cb(pjsip_inv_session *inv, pjsip_event *e);
void transaction_state_changed_cb(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
void registration_cb(pjsip_regc_cbparam *param);
pj_bool_t transaction_request_cb(pjsip_rx_data *rdata);
pj_bool_t transaction_response_cb(pjsip_rx_data *rdata) ;
void transfer_client_cb(pjsip_evsub *sub, pjsip_event *event);
/**
* Send a reINVITE inside an active dialog to modify its state
* Local SDP session should be modified before calling this method
* @param sip call
*/
int SIPSessionReinvite(SIPCall *);
/**
* Helper function to process refer function on call transfer
*/
void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata);
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);
}
// Always return PJ_TRUE since we are the only module that will handle these requests
pj_bool_t transaction_response_cb(pjsip_rx_data *rdata)
{
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
if (!dlg)
return PJ_TRUE;
pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
if (!tsx or tsx->method.id != PJSIP_INVITE_METHOD)
return PJ_TRUE;
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_TRUE;
}
// Always return PJ_TRUE since we are the only module that will handle these requests
pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
{
if (!rdata or !rdata->msg_info.msg) {
ERROR("rx_data is NULL");
return PJ_TRUE;
}
pjsip_method *method = &rdata->msg_info.msg->line.req.method;
if (!method) {
ERROR("method is NULL");
return PJ_TRUE;
}
if (method->id == PJSIP_ACK_METHOD && pjsip_rdata_get_dlg(rdata))
return PJ_TRUE;
if (!rdata->msg_info.to or !rdata->msg_info.from) {
ERROR("NULL from/to fields");
return PJ_TRUE;
}
pjsip_sip_uri *sip_to_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(rdata->msg_info.to->uri);
pjsip_sip_uri *sip_from_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(rdata->msg_info.from->uri);
if (!sip_to_uri or !sip_from_uri) {
ERROR("NULL uri");
return PJ_TRUE;
}
std::string userName(sip_to_uri->user.ptr, sip_to_uri->user.slen);
std::string server(sip_from_uri->host.ptr, sip_from_uri->host.slen);
std::string account_id(Manager::instance().getAccountIdFromNameAndServer(userName, server));
std::string displayName(sip_utils::parseDisplayName(rdata->msg_info.msg_buf));
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);
}
}
pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
return PJ_TRUE;
} else if (method->id == PJSIP_OPTIONS_METHOD) {
handleIncomingOptions(rdata);
return PJ_TRUE;
} else if (method->id != PJSIP_INVITE_METHOD && method->id != PJSIP_ACK_METHOD) {
pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return PJ_TRUE;
}
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
if (!account) {
ERROR("Could not find account %s", account_id.c_str());
return PJ_TRUE;
}
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 (account->getActiveAudioCodecs().empty()) {
pjsip_endpt_respond_stateless(endpt_, rdata,
PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL,
NULL);
return PJ_TRUE;
}
// Verify that we can handle the request
unsigned options = 0;
if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, endpt_, NULL) != PJ_SUCCESS) {
pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return PJ_TRUE;
}
Manager::instance().hookPreference.runHook(rdata->msg_info.msg);
SIPCall* call = new SIPCall(Manager::instance().getNewCallID(), Call::INCOMING, cp_);
Manager::instance().associateCallToAccount(call->getCallId(), account_id);
// May use the published address as well
std::string addrToUse = SipTransport::getInterfaceAddrFromName(account->getLocalInterface());
std::string addrSdp = account->isStunEnabled()
? account->getPublishedAddress()
: addrToUse;
pjsip_tpselector *tp = SIPVoIPLink::instance()->sipTransport.initTransportSelector(account->transport_, call->getMemoryPool());
if (addrToUse == "0.0.0.0")
addrToUse = SipTransport::getSIPLocalIP();
if (addrSdp == "0.0.0.0")
addrSdp = addrToUse;
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, std::min(length, sizeof tmp));
sip_utils::stripSipUriPrefix(peerNumber);
call->setConnectionState(Call::PROGRESSING);
call->setPeerNumber(peerNumber);
call->setDisplayName(displayName);
call->initRecFilename(peerNumber);
setCallMediaLocal(call, addrToUse);
call->getLocalSDP()->setLocalIP(addrSdp);
call->getAudioRtp().initConfig();
call->getAudioRtp().initSession();
if (body and body->len > 0) {
std::string sdpOffer(static_cast<const char*>(body->data), body->len);
size_t start = sdpOffer.find("a=crypto:");
// Found crypto header in SDP
if (start != std::string::npos) {
CryptoOffer crypto_offer;
crypto_offer.push_back(std::string(sdpOffer.substr(start, (sdpOffer.size() - start) - 1)));
const size_t size = ARRAYSIZE(sfl::CryptoSuites);
std::vector<sfl::CryptoSuiteDefinition> localCapabilities(size);
std::copy(sfl::CryptoSuites, sfl::CryptoSuites + size,
localCapabilities.begin());
sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
if (sdesnego.negotiate()) {
call->getAudioRtp().setRemoteCryptoInfo(sdesnego);
call->getAudioRtp().initLocalCryptoInfo();
}
}
}
#ifdef SFL_VIDEO
call->getLocalSDP()->receiveOffer(r_sdp, account->getActiveAudioCodecs(), account->getActiveVideoCodecs());
#else
call->getLocalSDP()->receiveOffer(r_sdp, account->getActiveAudioCodecs());
#endif
sfl::AudioCodec* ac = dynamic_cast<sfl::AudioCodec*>(Manager::instance().audioCodecFactory.instantiateCodec(PAYLOAD_CODEC_ULAW));
if (!ac) {
ERROR("Could not instantiate codec");
delete call;
return PJ_TRUE;
}
call->getAudioRtp().start(ac);
pjsip_dialog *dialog = 0;
if (pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS) {
delete call;
pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
return PJ_TRUE;
}
pjsip_inv_create_uas(dialog, rdata, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv);
if (pjsip_dlg_set_transport(dialog, tp) != PJ_SUCCESS) {
ERROR("Could not set transport for dialog");
delete call;
return PJ_TRUE;
}
if (!call->inv) {
ERROR("Call invite is not initialized");
delete call;
return PJ_TRUE;
}
call->inv->mod_data[mod_ua_.id] = call;
// 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) {
ERROR("Something wrong with Replaces request.");
delete call;
pjsip_endpt_respond_stateless(endpt_, rdata, 500 /* internal server error */, NULL, NULL, NULL);
}
// Check if call has been transfered
pjsip_tx_data *tdata = 0;
// If Replace header present
if (replaced_dlg) {
// Always answer the new INVITE with 200, regardless whether
// the replaced call is in early or confirmed state.
if (pjsip_inv_answer(call->inv, 200, NULL, NULL, &response) == PJ_SUCCESS)
pjsip_inv_send_msg(call->inv, response);
// Get the INVITE session associated with the replaced dialog.
pjsip_inv_session *replaced_inv = pjsip_dlg_get_inv_session(replaced_dlg);
// Disconnect the "replaced" INVITE session.
if (pjsip_inv_end_session(replaced_inv, PJSIP_SC_GONE, NULL, &tdata) == PJ_SUCCESS && tdata)
pjsip_inv_send_msg(replaced_inv, tdata);
} else { // Prooceed with normal call flow
if (pjsip_inv_initial_answer(call->inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata) != PJ_SUCCESS) {
ERROR("Could not answer invite");
delete call;
return PJ_TRUE;
}
if (pjsip_inv_send_msg(call->inv, tdata) != PJ_SUCCESS) {
ERROR("Could not send msg for invite");
delete call;
return PJ_TRUE;
}
call->setConnectionState(Call::RINGING);
Manager::instance().incomingCall(*call, account_id);
Manager::instance().getAccountLink(account_id)->addCall(call);
}
return PJ_TRUE;
}
} // end anonymous namespace
/*************************************************************************************************/
SIPVoIPLink::SIPVoIPLink() : sipTransport(endpt_, cp_, pool_), evThread_(this)
{
#define TRY(ret) do { \
if (ret != PJ_SUCCESS) \
throw VoipLinkException(#ret " failed"); \
} while (0)
srand(time(NULL)); // to get random number for RANDOM_PORT
TRY(pj_init());
TRY(pjlib_util_init());
// From 0 (min) to 6 (max)
// pj_log_set_level(Logger::getDebugMode() ? 6 : 0);
pj_log_set_level(0);
TRY(pjnath_init());
pj_caching_pool_init(cp_, &pj_pool_factory_default_policy, 0);
pool_ = pj_pool_create(&cp_->factory, "sflphone", 4000, 4000, NULL);
if (!pool_)
throw VoipLinkException("UserAgent: Could not initialize memory pool");
TRY(pjsip_endpt_create(&cp_->factory, pj_gethostname()->ptr, &endpt_));
sipTransport.setEndpoint(endpt_);
sipTransport.setCachingPool(cp_);
sipTransport.setPool(pool_);
if (SipTransport::getSIPLocalIP().empty())
throw VoipLinkException("UserAgent: Unable to determine network capabilities");
TRY(pjsip_tsx_layer_init_module(endpt_));
TRY(pjsip_ua_init_module(endpt_, NULL));
TRY(pjsip_replaces_init_module(endpt_)); // See the Replaces specification in RFC 3891
TRY(pjsip_100rel_init_module(endpt_));
// Initialize and register sflphone 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_));
static const pjsip_inv_callback inv_cb = {
invite_session_state_changed_cb,
outgoing_request_forked_cb,
transaction_state_changed_cb,
sdp_request_offer_cb,
sdp_create_offer_cb,
sdp_media_update_cb,
NULL,
NULL,
};
TRY(pjsip_inv_usage_init(endpt_, &inv_cb));
static const pj_str_t allowed[] = { { (char*) "INFO", 4}, { (char*) "REGISTER", 8}, { (char*) "OPTIONS", 7}, { (char*) "MESSAGE", 7 } }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
static const pj_str_t text_plain = { (char*) "text/plain", 10 };
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, NULL, 1, &text_plain);
static const pj_str_t accepted = { (char*) "application/sdp", 15 };
pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, NULL, 1, &accepted);
DEBUG("pjsip version %s for %s initialized", pj_get_version(), PJ_OS_NAME);
TRY(pjsip_replaces_init_module(endpt_));
#undef TRY
handlingEvents_ = true;
evThread_.start();
}
SIPVoIPLink::~SIPVoIPLink()
{
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);
handlingEvents_ = false;
if (thread_) {
pj_thread_join(thread_);
pj_thread_destroy(thread_);
DEBUG("PJ thread destroy finished");
thread_ = 0;
}
const pj_time_val tv = {0, 10};
pjsip_endpt_handle_events(endpt_, &tv);
pjsip_endpt_destroy(endpt_);
pj_pool_release(pool_);
pj_caching_pool_destroy(cp_);
pj_shutdown();
}
SIPVoIPLink* SIPVoIPLink::instance()
{
assert(!destroyed_);
if (!instance_)
instance_ = new SIPVoIPLink;
return instance_;
}
void SIPVoIPLink::destroy()
{
delete instance_;
destroyed_ = true;
instance_ = 0;
}
// Called from EventThread::run (not main thread)
bool SIPVoIPLink::getEvent()
{
static pj_thread_desc desc;
// We have to register the external thread so it could access the pjsip frameworks
if (!pj_thread_is_registered()) {
DEBUG("Registering thread");
pj_thread_register(NULL, desc, &thread_);
}
static const pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(endpt_, &timeout);
return handlingEvents_;
}
void SIPVoIPLink::sendRegister(Account *a)
{
SIPAccount *account = dynamic_cast<SIPAccount*>(a);
if (!account)
throw VoipLinkException("SipVoipLink: Account is not SIPAccount");
sipTransport.createSipTransport(*account);
account->setRegister(true);
account->setRegistrationState(Trying);
pjsip_regc *regc = account->getRegistrationInfo();
if (pjsip_regc_create(endpt_, (void *) account, &registration_cb, &regc) != PJ_SUCCESS)
throw VoipLinkException("UserAgent: Unable to create regc structure.");
std::string srvUri(account->getServerUri());
// std::string address, port;
// findLocalAddressFromUri(srvUri, account->transport_, address, port);
pj_str_t pjSrv = pj_str((char*) srvUri.c_str());
// Generate the FROM header
std::string from(account->getFromUri());
pj_str_t pjFrom = pj_str((char*) from.c_str());
// Get the received header
std::string received(account->getReceivedParameter());
// Get the contact header
std::string contact = account->getContactHeader();
pj_str_t pjContact = pj_str((char*) contact.c_str());
if (!received.empty()) {
// Set received parameter string to empty in order to avoid creating new transport for each register
account->setReceivedParameter("");
// Explicitely set the bound address port to 0 so that pjsip determine a random port by itself
account->transport_= sipTransport.createUdpTransport(account->getLocalInterface(), 0, received, account->getRPort());
account->setRPort(-1);
if(account->transport_ == NULL) {
ERROR("Could not create new udp transport with public address: %s:%d", received.c_str(), account->getLocalPort());
}
}
if (pjsip_regc_init(regc, &pjSrv, &pjFrom, &pjFrom, 1, &pjContact, account->getRegistrationExpire()) != PJ_SUCCESS)
throw VoipLinkException("Unable to initialize account registration structure");
if (not account->getServiceRoute().empty())
pjsip_regc_set_route_set(regc, sip_utils::createRouteSet(account->getServiceRoute(), pool_));
pjsip_regc_set_credentials(regc, account->getCredentialCount(), account->getCredInfo());
pjsip_hdr hdr_list;
pj_list_init(&hdr_list);
std::string useragent(account->getUserAgentName());
pj_str_t pJuseragent = pj_str((char*) useragent.c_str());
const pj_str_t STR_USER_AGENT = { (char*) "User-Agent", 10 };
pjsip_generic_string_hdr *h = pjsip_generic_string_hdr_create(pool_, &STR_USER_AGENT, &pJuseragent);
pj_list_push_back(&hdr_list, (pjsip_hdr*) h);
pjsip_regc_add_headers(regc, &hdr_list);
pjsip_tx_data *tdata;
if (pjsip_regc_register(regc, PJ_TRUE, &tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to initialize transaction data for account registration");
if (pjsip_regc_set_transport(regc, sipTransport.initTransportSelector(account->transport_, pool_)) != PJ_SUCCESS)
throw VoipLinkException("Unable to set transport");
// decrease transport's ref count, counter incrementation is managed when acquiring transport
pjsip_transport_dec_ref(account->transport_);
// pjsip_regc_send increment the transport ref count by one,
if (pjsip_regc_send(regc, tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to send account registration request");
// Decrease transport's ref count, since coresponding reference counter decrementation
// is performed in pjsip_regc_destroy. This function is never called in SFLphone as the
// regc data structure is permanently associated to the account at first registration.
pjsip_transport_dec_ref(account->transport_);
account->setRegistrationInfo(regc);
// start the periodic registration request based on Expire header
// account determines itself if a keep alive is required
if(account->isKeepAliveEnabled())
account->startKeepAliveTimer();
}
void SIPVoIPLink::sendUnregister(Account *a)
{
SIPAccount *account = dynamic_cast<SIPAccount *>(a);
// This may occurs if account failed to register and is in state INVALID
if (!account->isRegistered()) {
account->setRegistrationState(Unregistered);
return;
}
// Make sure to cancel any ongoing timers before unregister
account->stopKeepAliveTimer();
pjsip_regc *regc = account->getRegistrationInfo();
if (!regc)
throw VoipLinkException("Registration structure is NULL");
pjsip_tx_data *tdata = NULL;
if (pjsip_regc_unregister(regc, &tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to unregister sip account");
if (pjsip_regc_send(regc, tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to send request to unregister sip account");
account->setRegister(false);
}
void SIPVoIPLink::registerKeepAliveTimer(pj_timer_entry &timer, pj_time_val &delay)
{
DEBUG("Register new keep alive timer %d with delay %d", timer.id, delay.sec);
if (timer.id == -1)
WARN("Timer already scheduled");
switch (pjsip_endpt_schedule_timer(endpt_, &timer, &delay)) {
case PJ_SUCCESS:
break;
default:
ERROR("Could not schedule new timer in pjsip endpoint");
/* fallthrough */
case PJ_EINVAL:
ERROR("Invalid timer or delay entry");
break;
case PJ_EINVALIDOP:
ERROR("Invalid timer entry, maybe already scheduled");
break;
}
}
void SIPVoIPLink::cancelKeepAliveTimer(pj_timer_entry& timer)
{
pjsip_endpt_cancel_timer(endpt_, &timer);
}
bool isValidIpAddress(const std::string &address)
{
size_t pos = address.find(":");
std::string address_without_port(address);
if (pos != std::string::npos)
address_without_port = address.substr(0, pos);
DEBUG("Testing address %s", address_without_port.c_str());
struct sockaddr_in sa;
int result = inet_pton(AF_INET, address_without_port.data(), &(sa.sin_addr));
return result != 0;
}
Call *SIPVoIPLink::newOutgoingCall(const std::string& id, const std::string& toUrl)
{
DEBUG("New outgoing call to %s", toUrl.c_str());
std::string toCpy = toUrl;
sip_utils::stripSipUriPrefix(toCpy);
const bool IPToIP = isValidIpAddress(toCpy);
Manager::instance().setIPToIPForCall(id, IPToIP);
if (IPToIP) {
Manager::instance().associateCallToAccount(id, SIPAccount::IP2IP_PROFILE);
return SIPNewIpToIpCall(id, toUrl);
} else {
return newRegisteredAccountCall(id, toUrl);
}
}
Call *SIPVoIPLink::SIPNewIpToIpCall(const std::string& id, const std::string& to)
{
DEBUG("New IP to IP call to %s", to.c_str());
SIPAccount *account = Manager::instance().getIP2IPAccount();
if (!account)
throw VoipLinkException("Could not retrieve default account for IP2IP call");
SIPCall *call = new SIPCall(id, Call::OUTGOING, cp_);
call->setIPToIP(true);
call->initRecFilename(to);
std::string localAddress(SipTransport::getInterfaceAddrFromName(account->getLocalInterface()));
if (localAddress == "0.0.0.0")
localAddress = SipTransport::getSIPLocalIP();
setCallMediaLocal(call, localAddress);
std::string toUri = account->getToUri(to);
call->setPeerNumber(toUri);
sfl::AudioCodec* ac = dynamic_cast<sfl::AudioCodec*>(Manager::instance().audioCodecFactory.instantiateCodec(PAYLOAD_CODEC_ULAW));
if (!ac) {
delete call;
throw VoipLinkException("Could not instantiate codec");
}
// Audio Rtp Session must be initialized before creating initial offer in SDP session
// since SDES require crypto attribute.
call->getAudioRtp().initConfig();
call->getAudioRtp().initSession();
call->getAudioRtp().initLocalCryptoInfo();
call->getAudioRtp().start(ac);
// Building the local SDP offer
call->getLocalSDP()->setLocalIP(localAddress);
#ifdef SFL_VIDEO
call->getLocalSDP()->createOffer(account->getActiveAudioCodecs(), account->getActiveVideoCodecs());
#else
call->getLocalSDP()->createOffer(account->getActiveAudioCodecs());
#endif
if (!SIPStartCall(call)) {
delete call;
throw VoipLinkException("Could not create new call");
}
return call;
}
Call *SIPVoIPLink::newRegisteredAccountCall(const std::string& id, const std::string& toUrl)
{
DEBUG("UserAgent: New registered account call to %s", toUrl.c_str());
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(Manager::instance().getAccountFromCall(id)));
if (account == NULL) // TODO: We should investigate how we could get rid of this error and create a IP2IP call instead
throw VoipLinkException("Could not get account for this call");
SIPCall* call = new SIPCall(id, Call::OUTGOING, cp_);
// If toUri is not a well formatted sip URI, use account information to process it
std::string toUri;
if (toUrl.find("sip:") != std::string::npos or
toUrl.find("sips:") != std::string::npos)
toUri = toUrl;
else
toUri = account->getToUri(toUrl);
call->setPeerNumber(toUri);
std::string localAddr(SipTransport::getInterfaceAddrFromName(account->getLocalInterface()));
if (localAddr == "0.0.0.0")
localAddr = SipTransport::getSIPLocalIP();
setCallMediaLocal(call, localAddr);
// May use the published address as well
std::string addrSdp = account->isStunEnabled() ?
account->getPublishedAddress() :
SipTransport::getInterfaceAddrFromName(account->getLocalInterface());
if (addrSdp == "0.0.0.0")
addrSdp = SipTransport::getSIPLocalIP();
// Initialize the session using ULAW as default codec in case of early media
// The session should be ready to receive media once the first INVITE is sent, before
// the session initialization is completed
sfl::AudioCodec* ac = dynamic_cast<sfl::AudioCodec*>(Manager::instance().audioCodecFactory.instantiateCodec(PAYLOAD_CODEC_ULAW));
if (ac == NULL) {
delete call;
throw VoipLinkException("Could not instantiate codec for early media");
}
try {
call->getAudioRtp().initConfig();
call->getAudioRtp().initSession();
call->getAudioRtp().initLocalCryptoInfo();
call->getAudioRtp().start(ac);
} catch (...) {
delete call;
throw VoipLinkException("Could not start rtp session for early media");
}
call->initRecFilename(toUrl);
call->getLocalSDP()->setLocalIP(addrSdp);
#ifdef SFL_VIDEO
call->getLocalSDP()->createOffer(account->getActiveAudioCodecs(), account->getActiveVideoCodecs());
#else
call->getLocalSDP()->createOffer(account->getActiveAudioCodecs());
#endif
if (!SIPStartCall(call)) {
delete call;
throw VoipLinkException("Could not send outgoing INVITE request for new call");
}
return call;
}
void
SIPVoIPLink::answer(Call *call)
{
if (!call)
return;
call->answer();
}
void
SIPVoIPLink::hangup(const std::string& id)
{
SIPCall* call = getSIPCall(id);
std::string account_id(Manager::instance().getAccountFromCall(id));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
if (account == NULL)
throw VoipLinkException("Could not find account for this call");
pjsip_inv_session *inv = call->inv;
if (inv == NULL)
throw VoipLinkException("No invite session for this call");
// Looks for sip routes
if (not account->getServiceRoute().empty()) {
pjsip_route_hdr *route_set = sip_utils::createRouteSet(account->getServiceRoute(), inv->pool);
pjsip_dlg_set_route_set(inv->dlg, route_set);
}
pjsip_tx_data *tdata = NULL;
// User hangup current call. Notify peer
if (pjsip_inv_end_session(inv, 404, NULL, &tdata) != PJ_SUCCESS || !tdata)
return;
if (pjsip_inv_send_msg(inv, tdata) != PJ_SUCCESS)
return;
// Make sure user data is NULL in callbacks
inv->mod_data[mod_ua_.id] = NULL;
if (Manager::instance().isCurrentCall(id))
call->getAudioRtp().stop();
removeCall(id);
}
void
SIPVoIPLink::peerHungup(const std::string& id)
{
SIPCall* call = getSIPCall(id);
// User hangup current call. Notify peer
pjsip_tx_data *tdata = NULL;
if (pjsip_inv_end_session(call->inv, 404, NULL, &tdata) != PJ_SUCCESS || !tdata)
return;
if (pjsip_inv_send_msg(call->inv, tdata) != PJ_SUCCESS)
return;
// Make sure user data is NULL in callbacks
call->inv->mod_data[mod_ua_.id ] = NULL;
if (Manager::instance().isCurrentCall(id))
call->getAudioRtp().stop();
removeCall(id);
}
void
SIPVoIPLink::onhold(const std::string& id)
{
SIPCall *call = getSIPCall(id);
call->setState(Call::HOLD);
call->getAudioRtp().saveLocalContext();
call->getAudioRtp().stop();
Sdp *sdpSession = call->getLocalSDP();
if (!sdpSession)
throw VoipLinkException("Could not find sdp session");
sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
sdpSession->addAttributeToLocalAudioMedia("sendonly");
SIPSessionReinvite(call);
}
void
SIPVoIPLink::offhold(const std::string& id)
{
SIPCall *call = getSIPCall(id);
Sdp *sdpSession = call->getLocalSDP();
if (sdpSession == NULL)
throw VoipLinkException("Could not find sdp session");
try {
int pl = PAYLOAD_CODEC_ULAW;
sfl::Codec *sessionMedia = sdpSession->getSessionAudioMedia();
if (sessionMedia)
pl = sessionMedia->getPayloadType();
// Create a new instance for this codec
sfl::AudioCodec* ac = dynamic_cast<sfl::AudioCodec*>(Manager::instance().audioCodecFactory.instantiateCodec(pl));
if (ac == NULL)
throw VoipLinkException("Could not instantiate codec");
call->getAudioRtp().initConfig();
call->getAudioRtp().initSession();
call->getAudioRtp().restoreLocalContext();
call->getAudioRtp().initLocalCryptoInfoOnOffHold();
call->getAudioRtp().start(ac);
} catch (const SdpException &e) {
ERROR("%s", e.what());
} catch (...) {
throw VoipLinkException("Could not create audio rtp session");
}
sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
sdpSession->addAttributeToLocalAudioMedia("sendrecv");
if (SIPSessionReinvite(call) == PJ_SUCCESS)
call->setState(Call::ACTIVE);
}
void SIPVoIPLink::sendTextMessage(const std::string &callID,
const std::string &message,
const std::string &from)
{
using namespace sfl::InstantMessaging;
SIPCall *call;
try {
call = getSIPCall(callID);
} catch (const VoipLinkException &e) {
return;
}
/* Send IM message */
UriList list;
UriEntry entry;
entry[sfl::IM_XML_URI] = std::string("\"" + from + "\""); // add double quotes for xml formating
list.push_front(entry);
send_sip_message(call->inv, callID, appendUriList(message, list));
}
bool
SIPVoIPLink::transferCommon(SIPCall *call, pj_str_t *dst)
{
if (!call or !call->inv)
return false;
pjsip_evsub_user xfer_cb;
pj_bzero(&xfer_cb, sizeof(xfer_cb));
xfer_cb.on_evsub_state = &transfer_client_cb;
pjsip_evsub *sub;
if (pjsip_xfer_create_uac(call->inv->dlg, &xfer_cb, &sub) != PJ_SUCCESS)
return false;
/* Associate this voiplink of call with the client subscription
* We can not just associate call with the client subscription
* because after this function, we can no find the cooresponding
* voiplink from the call any more. But the voiplink is useful!
*/
pjsip_evsub_set_mod_data(sub, mod_ua_.id, this);
/*
* Create REFER request.
*/
pjsip_tx_data *tdata;
if (pjsip_xfer_initiate(sub, dst, &tdata) != PJ_SUCCESS)
return false;
// Put SIP call id in map in order to retrieve call during transfer callback
std::string callidtransfer(call->inv->dlg->call_id->id.ptr, call->inv->dlg->call_id->id.slen);
transferCallID[callidtransfer] = call->getCallId();
/* Send. */
if (pjsip_xfer_send_request(sub, tdata) != PJ_SUCCESS)
return false;
return true;
}
void
SIPVoIPLink::transfer(const std::string& id, const std::string& to)
{
SIPCall *call = getSIPCall(id);
if (call == NULL) {
ERROR("Could not find call %s", id.c_str());
return;
}
call->stopRecording();
std::string account_id(Manager::instance().getAccountFromCall(id));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
if (account == NULL)
throw VoipLinkException("Could not find account");
std::string toUri;
pj_str_t dst = { 0, 0 };
if (to.find("@") == std::string::npos) {
toUri = account->getToUri(to);
pj_cstr(&dst, toUri.c_str());
}
if (!transferCommon(getSIPCall(id), &dst))
throw VoipLinkException("Couldn't transfer");
}
bool SIPVoIPLink::attendedTransfer(const std::string& id, const std::string& to)
{
SIPCall *call = getSIPCall(to);
if (!call or !call->inv or !call->inv->dlg)
throw VoipLinkException("Couldn't get invite dialog");
pjsip_dialog *target_dlg = call->inv->dlg;
pjsip_uri *uri = (pjsip_uri*) pjsip_uri_get_uri(target_dlg->remote.info->uri);
char str_dest_buf[PJSIP_MAX_URL_SIZE * 2] = { '<' };
pj_str_t dst = { str_dest_buf, 1 };
dst.slen += pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, str_dest_buf+1, sizeof(str_dest_buf)-1);
dst.slen += pj_ansi_snprintf(str_dest_buf + dst.slen,
sizeof(str_dest_buf) - dst.slen,
"?"
"Replaces=%.*s"
"%%3Bto-tag%%3D%.*s"
"%%3Bfrom-tag%%3D%.*s>",
(int)target_dlg->call_id->id.slen,
target_dlg->call_id->id.ptr,
(int)target_dlg->remote.info->tag.slen,
target_dlg->remote.info->tag.ptr,
(int)target_dlg->local.info->tag.slen,
target_dlg->local.info->tag.ptr);
return transferCommon(getSIPCall(id), &dst);
}
void
SIPVoIPLink::refuse(const std::string& id)
{
SIPCall *call = getSIPCall(id);
if (!call or !call->isIncoming() or call->getConnectionState() == Call::CONNECTED or !call->inv)
return;
call->getAudioRtp().stop();
pjsip_tx_data *tdata;
if (pjsip_inv_end_session(call->inv, PJSIP_SC_DECLINE, NULL, &tdata) != PJ_SUCCESS)
return;
if (pjsip_inv_send_msg(call->inv, tdata) != PJ_SUCCESS)
return;
// Make sure the pointer is NULL in callbacks
call->inv->mod_data[mod_ua_.id] = NULL;
removeCall(id);
}
#ifdef SFL_VIDEO
std::string
SIPVoIPLink::getCurrentVideoCodecName(const std::string& id)
{
SIPCall *call = getSIPCall(id);
if (call) {
Call::CallState state = call->getState();
if (state == Call::ACTIVE or state == Call::CONFERENCING)
return call->getLocalSDP()->getSessionVideoCodec();
}
return "";
}
#endif
std::string
SIPVoIPLink::getCurrentCodecName(Call *call) const
{
return dynamic_cast<SIPCall*>(call)->getLocalSDP()->getAudioCodecName();
}
void
SIPVoIPLink::carryingDTMFdigits(const std::string& id, char code)
{
std::string accountID(Manager::instance().getAccountFromCall(id));
SIPAccount *account = dynamic_cast<SIPAccount*>(Manager::instance().getAccount(accountID));
if (account) {
try {
dtmfSend(getSIPCall(id), code, account->getDtmfType());
} catch (const VoipLinkException &e) {
// don't do anything if call doesn't exist
}
}
}
void
SIPVoIPLink::dtmfSend(SIPCall *call, char code, const std::string &dtmf)
{
if (dtmf == SIPAccount::OVERRTP_STR) {
call->getAudioRtp().sendDtmfDigit(code - '0');
return;
}
else if (dtmf != SIPAccount::SIPINFO_STR) {
WARN("SIPVoIPLink: Unknown DTMF type %s, defaulting to %s instead",
dtmf.c_str(), SIPAccount::SIPINFO_STR);
}
// else : dtmf == SIPINFO
pj_str_t methodName = pj_str((char*) "INFO");
pjsip_method method;
pjsip_method_init_np(&method, &methodName);
/* Create request message. */
pjsip_tx_data *tdata;
if (pjsip_dlg_create_request(call->inv->dlg, &method, -1, &tdata) != PJ_SUCCESS)
return;
int duration = Manager::instance().voipPreferences.getPulseLength();
char dtmf_body[1000];
snprintf(dtmf_body, sizeof dtmf_body - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
/* Create "application/dtmf-relay" message body. */
pj_str_t content = pj_str(dtmf_body);
pj_str_t type = pj_str((char*) "application");
pj_str_t subtype = pj_str((char*) "dtmf-relay");
tdata->msg->body = pjsip_msg_body_create(tdata->pool, &type, &subtype, &content);
if (tdata->msg->body == NULL)
pjsip_tx_data_dec_ref(tdata);
else
pjsip_dlg_send_request(call->inv->dlg, tdata, mod_ua_.id, NULL);
}
bool
SIPVoIPLink::SIPStartCall(SIPCall *call)
{
std::string id(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
if (account == NULL) {
ERROR("Account is NULL in SIPStartCall");
return false;
}
std::string toUri(call->getPeerNumber()); // expecting a fully well formed sip uri
pj_str_t pjTo = pj_str((char*) toUri.c_str());
// Create the from header
std::string from(account->getFromUri());
pj_str_t pjFrom = pj_str((char*) from.c_str());
// Get the contact header
std::string contact(account->getContactHeader());
pj_str_t pjContact = pj_str((char*) contact.c_str());
pjsip_dialog *dialog = NULL;
if (pjsip_dlg_create_uac(pjsip_ua_instance(), &pjFrom, &pjContact, &pjTo, NULL, &dialog) != PJ_SUCCESS) {
ERROR("Unable to sip create dialogs for user agent client");
return false;
}
if (pjsip_inv_create_uac(dialog, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv) != PJ_SUCCESS) {
ERROR("Unable to create invite session for user agent client");
return false;
}
if (not account->getServiceRoute().empty())
pjsip_dlg_set_route_set(dialog, sip_utils::createRouteSet(account->getServiceRoute(), call->inv->pool));
if (pjsip_auth_clt_set_credentials(&dialog->auth_sess, account->getCredentialCount(), account->getCredInfo()) != PJ_SUCCESS) {
ERROR("Could not initialize credential for invite session authentication");
return false;
}
call->inv->mod_data[mod_ua_.id] = call;
pjsip_tx_data *tdata;
if (pjsip_inv_invite(call->inv, &tdata) != PJ_SUCCESS) {
ERROR("Could not initialize invite messager for this call");
return false;
}
pjsip_tpselector *tp = sipTransport.initTransportSelector(account->transport_, call->inv->pool);
if (pjsip_dlg_set_transport(dialog, tp) != PJ_SUCCESS) {
ERROR("Unable to associate transport fir invite session dialog");
return false;
}
if (pjsip_inv_send_msg(call->inv, tdata) != PJ_SUCCESS) {
ERROR("Unable to send invite message for this call");
return false;
}
call->setConnectionState(Call::PROGRESSING);
call->setState(Call::ACTIVE);
addCall(call);
return true;
}
void
SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
{
std::string id(call->getCallId());
Manager::instance().callFailure(id);
removeCall(id);
}
void
SIPVoIPLink::SIPCallClosed(SIPCall *call)
{
std::string id(call->getCallId());
if (Manager::instance().isCurrentCall(id))
call->getAudioRtp().stop();
Manager::instance().peerHungupCall(id);
removeCall(id);
}
void
SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data * /*rdata*/)
{
if (call->getConnectionState() != Call::CONNECTED) {
call->setConnectionState(Call::CONNECTED);
call->setState(Call::ACTIVE);
Manager::instance().peerAnsweredCall(call->getCallId());
}
}
SIPCall*
SIPVoIPLink::getSIPCall(const std::string& id)
{
SIPCall *result = dynamic_cast<SIPCall*>(getCall(id));
if (result == NULL)
throw VoipLinkException("Could not find SIPCall " + id);
return result;
}
///////////////////////////////////////////////////////////////////////////////
// Private functions
///////////////////////////////////////////////////////////////////////////////
namespace {
int SIPSessionReinvite(SIPCall *call)
{
pjmedia_sdp_session *local_sdp = call->getLocalSDP()->getLocalSdpSession();
pjsip_tx_data *tdata;
if (local_sdp && pjsip_inv_reinvite(call->inv, NULL, local_sdp, &tdata) == PJ_SUCCESS)
return pjsip_inv_send_msg(call->inv, tdata);
return !PJ_SUCCESS;
}
void invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *ev)
{
if (!inv)
return;
SIPCall *call = static_cast<SIPCall*>(inv->mod_data[mod_ua_.id]);
if (call == NULL)
return;
if (ev and inv->state != PJSIP_INV_STATE_CONFIRMED) {
// Update UI with the current status code and description
pjsip_transaction * tsx = ev->body.tsx_state.tsx;
int statusCode = tsx ? tsx->status_code : 404;
if (statusCode) {
const pj_str_t * description = pjsip_get_status_text(statusCode);
std::string desc(description->ptr, description->slen);
CallManager *cm = Manager::instance().getDbusManager()->getCallManager();
cm->sipCallStateChanged(call->getCallId(), desc, statusCode);
}
}
SIPVoIPLink *link = SIPVoIPLink::instance();
if (inv->state == PJSIP_INV_STATE_EARLY and ev and ev->body.tsx_state.tsx and
ev->body.tsx_state.tsx->role == PJSIP_ROLE_UAC) {
call->setConnectionState(Call::RINGING);
Manager::instance().peerRingingCall(call->getCallId());
} else if (inv->state == PJSIP_INV_STATE_CONFIRMED and ev) {
// After we sent or received a ACK - The connection is established
link->SIPCallAnswered(call, ev->body.tsx_state.src.rdata);
} else if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
std::string accId(Manager::instance().getAccountFromCall(call->getCallId()));
switch (inv->cause) {
// The call terminates normally - BYE / CANCEL
case PJSIP_SC_OK:
case PJSIP_SC_REQUEST_TERMINATED:
link->SIPCallClosed(call);
break;
case PJSIP_SC_DECLINE:
if (inv->role != PJSIP_ROLE_UAC)
break;
case PJSIP_SC_NOT_FOUND:
case PJSIP_SC_REQUEST_TIMEOUT:
case PJSIP_SC_NOT_ACCEPTABLE_HERE: /* no compatible codecs */
case PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE:
case PJSIP_SC_UNSUPPORTED_MEDIA_TYPE:
case PJSIP_SC_UNAUTHORIZED:
case PJSIP_SC_FORBIDDEN:
case PJSIP_SC_REQUEST_PENDING:
case PJSIP_SC_ADDRESS_INCOMPLETE:
default:
link->SIPCallServerFailure(call);
break;
}
}
}
void sdp_request_offer_cb(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
{
if (!inv)
return;
SIPCall *call = static_cast<SIPCall*>(inv->mod_data[mod_ua_.id]);
if (!call)
return;
std::string accId(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accId));
if (!account)
return;
#ifdef SFL_VIDEO
call->getLocalSDP()->receiveOffer(offer, account->getActiveAudioCodecs(), account->getActiveVideoCodecs());
#else
call->getLocalSDP()->receiveOffer(offer, account->getActiveAudioCodecs());
#endif
call->getLocalSDP()->startNegotiation();
pjsip_inv_set_sdp_answer(call->inv, call->getLocalSDP()->getLocalSdpSession());
}
void sdp_create_offer_cb(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
if (!inv or !p_offer)
return;
SIPCall *call = static_cast<SIPCall*>(inv->mod_data[mod_ua_.id]);
if (!call)
return;
std::string accountid(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accountid));
std::string localAddress(SipTransport::getInterfaceAddrFromName(account->getLocalInterface()));
std::string addrSdp(localAddress);
if (localAddress == "0.0.0.0")
localAddress = SipTransport::getSIPLocalIP();
if (addrSdp == "0.0.0.0")
addrSdp = localAddress;
setCallMediaLocal(call, localAddress);
call->getLocalSDP()->setLocalIP(addrSdp);
#ifdef SFL_VIDEO
call->getLocalSDP()->createOffer(account->getActiveAudioCodecs(), account->getActiveVideoCodecs());
#else
call->getLocalSDP()->createOffer(account->getActiveAudioCodecs());
#endif
*p_offer = call->getLocalSDP()->getLocalSdpSession();
}
// This callback is called after SDP offer/answer session has completed.
void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
{
if (!inv)
return;
SIPCall *call = static_cast<SIPCall *>(inv->mod_data[mod_ua_.id]);
if (call == NULL) {
DEBUG("Call declined by peer, SDP negotiation stopped");
return;
}
if (status != PJ_SUCCESS) {
WARN("Could not negotiate offer");
SIPVoIPLink::instance()->hangup(call->getCallId());
Manager::instance().callFailure(call->getCallId());
return;
}
if (!inv->neg) {
WARN("No negotiator for this session");
return;
}
// Retreive SDP session for this call
Sdp *sdpSession = call->getLocalSDP();
if (!sdpSession) {
ERROR("No SDP session");
return;
}
// Get active session sessions
const pjmedia_sdp_session *remote_sdp = 0;
pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
if (pjmedia_sdp_validate(remote_sdp) != PJ_SUCCESS) {
ERROR("Invalid remote SDP session");
return;
}
const pjmedia_sdp_session *local_sdp;
pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
if (pjmedia_sdp_validate(local_sdp) != PJ_SUCCESS) {
ERROR("Invalid local SDP session");
return;
}
// Print SDP session
char buffer[4096];
memset(buffer, 0, sizeof buffer);
if (pjmedia_sdp_print(remote_sdp, buffer, sizeof buffer) == -1) {
ERROR("SDP was too big for buffer");
return;
}
DEBUG("Remote active SDP Session:\n%s", buffer);
memset(buffer, 0, sizeof buffer);
if (pjmedia_sdp_print(local_sdp, buffer, sizeof buffer) == -1) {
ERROR("SDP was too big for buffer");
return;
}
DEBUG("Local active SDP Session:\n%s", buffer);
// Set active SDP sessions
sdpSession->setActiveRemoteSdpSession(remote_sdp);
sdpSession->setActiveLocalSdpSession(local_sdp);
// Update internal field for
sdpSession->setMediaTransportInfoFromRemoteSdp();
call->getAudioRtp().updateDestinationIpAddress();
call->getAudioRtp().setDtmfPayloadType(sdpSession->getTelephoneEventType());
#ifdef SFL_VIDEO
call->getVideoRtp().updateSDP(*call->getLocalSDP());
call->getVideoRtp().updateDestination(call->getLocalSDP()->getRemoteIP(), call->getLocalSDP()->getRemoteVideoPort());
call->getVideoRtp().start();
#endif
// Get the crypto attribute containing srtp's cryptographic context (keys, cipher)
CryptoOffer crypto_offer;
call->getLocalSDP()->getRemoteSdpCryptoFromOffer(remote_sdp, crypto_offer);
bool nego_success = false;
if (!crypto_offer.empty()) {
std::vector<sfl::CryptoSuiteDefinition> localCapabilities;
for (size_t i = 0; i < ARRAYSIZE(sfl::CryptoSuites); ++i)
localCapabilities.push_back(sfl::CryptoSuites[i]);
sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
if (sdesnego.negotiate()) {
nego_success = true;
try {
call->getAudioRtp().setRemoteCryptoInfo(sdesnego);
} catch (...) {}
Manager::instance().getDbusManager()->getCallManager()->secureSdesOn(call->getCallId());
} else {
ERROR("SDES negotiation failure");
Manager::instance().getDbusManager()->getCallManager()->secureSdesOff(call->getCallId());
}
}
else {
DEBUG("No crypto offer available");
}
// We did not find any crypto context for this media, RTP fallback
if (!nego_success && call->getAudioRtp().isSdesEnabled()) {
ERROR("Negotiation failed but SRTP is enabled, fallback on RTP");
call->getAudioRtp().stop();
call->getAudioRtp().setSrtpEnabled(false);
std::string accountID = Manager::instance().getAccountFromCall(call->getCallId());
if (dynamic_cast<SIPAccount*>(Manager::instance().getAccount(accountID))->getSrtpFallback())
call->getAudioRtp().initSession();
}
sfl::AudioCodec *sessionMedia = sdpSession->getSessionAudioMedia();
if (!sessionMedia)
return;
try {
Manager::instance().audioLayerMutexLock();
Manager::instance().getAudioDriver()->startStream();
Manager::instance().audioLayerMutexUnlock();
int pl = sessionMedia->getPayloadType();
if (pl != call->getAudioRtp().getSessionMedia()) {
sfl::AudioCodec *ac = dynamic_cast<sfl::AudioCodec*>(Manager::instance().audioCodecFactory.instantiateCodec(pl));
if (!ac)
throw std::runtime_error("Could not instantiate codec");
call->getAudioRtp().updateSessionMedia(ac);
}
} catch (const SdpException &e) {
ERROR("%s", e.what());
} catch (const std::exception &rtpException) {
ERROR("%s", rtpException.what());
}
}
void outgoing_request_forked_cb(pjsip_inv_session * /*inv*/, pjsip_event * /*e*/)
{}
void transaction_state_changed_cb(pjsip_inv_session * inv,
pjsip_transaction *tsx, pjsip_event *event)
{
if (!tsx or !event or !inv or tsx->role != PJSIP_ROLE_UAS or
tsx->state != PJSIP_TSX_STATE_TRYING)
return;
// Handle the refer method
if (pjsip_method_cmp(&tsx->method, &pjsip_refer_method) == 0) {
onCallTransfered(inv, event->body.tsx_state.src.rdata);
return;
}
pjsip_tx_data* t_data;
if (event->body.rx_msg.rdata) {
pjsip_rx_data *r_data = event->body.rx_msg.rdata;
if (r_data && r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD) {
std::string request = pjsip_rx_data_get_info(r_data);
DEBUG("%s", request.c_str());
if (request.find("NOTIFY") == std::string::npos && request.find("INFO") != std::string::npos) {
pjsip_dlg_create_response(inv->dlg, r_data, PJSIP_SC_OK, NULL, &t_data);
pjsip_dlg_send_response(inv->dlg, tsx, t_data);
return;
}
}
}
if (!event->body.tsx_state.src.rdata)
return;
// Incoming TEXT message
// Get the message inside the transaction
pjsip_rx_data *r_data = event->body.tsx_state.src.rdata;
if (!r_data->msg_info.msg->body)
return;
const char *formattedMsgPtr = static_cast<const char*>(r_data->msg_info.msg->body->data);
if (!formattedMsgPtr)
return;
std::string formattedMessage(formattedMsgPtr, strlen(formattedMsgPtr));
// Try to determine who is the recipient of the message
SIPCall *call = static_cast<SIPCall *>(inv->mod_data[mod_ua_.id]);
if (!call)
return;
// Respond with a 200/OK
pjsip_dlg_create_response(inv->dlg, r_data, PJSIP_SC_OK, NULL, &t_data);
pjsip_dlg_send_response(inv->dlg, tsx, t_data);
using namespace sfl::InstantMessaging;
try {
// retreive the recipient-list of this message
std::string urilist = findTextUriList(formattedMessage);
UriList list = parseXmlUriList(urilist);
// If no item present in the list, peer is considered as the sender
std::string from;
if (list.empty()) {
from = call->getPeerNumber();
} else {
from = list.front()[IM_XML_URI];
if (from == "Me")
from = call->getPeerNumber();
}
// strip < and > characters in case of an IP address
if (from[0] == '<' && from[from.size()-1] == '>')
from = from.substr(1, from.size()-2);
Manager::instance().incomingMessage(call->getCallId(), from, findTextMessage(formattedMessage));
} catch (const sfl::InstantMessageException &except) {
ERROR("%s", except.what());
}
}
void update_contact_header(pjsip_regc_cbparam *param, SIPAccount *account)
{
SIPVoIPLink *siplink = dynamic_cast<SIPVoIPLink *>(account->getVoIPLink());
if (siplink == NULL) {
ERROR("Could not find voip link from account");
return;
}
pj_pool_t *pool = pj_pool_create(&cp_->factory, "tmp", 512, 512, NULL);
if (pool == NULL) {
ERROR("Could not create temporary memory pool in transport header");
return;
}
if (!param or param->contact_cnt == 0) {
WARN("SIPVoIPLink: No contact header in registration callback");
pj_pool_release(pool);
return;
}
pjsip_contact_hdr *contact_hdr = param->contact[0];
if (!contact_hdr)
return;
pjsip_sip_uri *uri = (pjsip_sip_uri*) contact_hdr->uri;
if (uri == NULL) {
ERROR("Could not find uri in contact header");
pj_pool_release(pool);
return;
}
// TODO: make this based on transport type
// with pjsip_transport_get_default_port_for_type(tp_type);
if (uri->port == 0) {
ERROR("Port is 0 in uri");
uri->port = DEFAULT_SIP_PORT;
}
std::string recvContactHost(uri->host.ptr, uri->host.slen);
std::stringstream ss;
ss << uri->port;
std::string recvContactPort = ss.str();
std::string currentAddress, currentPort;
siplink->sipTransport.findLocalAddressFromTransport(account->transport_, PJSIP_TRANSPORT_UDP, currentAddress, currentPort);
bool updateContact = false;
std::string currentContactHeader = account->getContactHeader();
size_t foundHost = currentContactHeader.find(recvContactHost);
if (foundHost == std::string::npos)
updateContact = true;
size_t foundPort = currentContactHeader.find(recvContactPort);
if (foundPort == std::string::npos)
updateContact = true;
if (updateContact) {
DEBUG("Update contact header: %s:%s\n", recvContactHost.c_str(), recvContactPort.c_str());
account->setContactHeader(recvContactHost, recvContactPort);
siplink->sendRegister(account);
}
pj_pool_release(pool);
}
void lookForReceivedParameter(pjsip_regc_cbparam &param, SIPAccount &account)
{
if (!param.rdata or !param.rdata->msg_info.via)
return;
pj_str_t receivedValue = param.rdata->msg_info.via->recvd_param;
if (receivedValue.slen) {
std::string publicIpFromReceived(receivedValue.ptr, receivedValue.slen);
account.setReceivedParameter(publicIpFromReceived);
}
account.setRPort(param.rdata->msg_info.via->rport_param);
}
void processRegistrationError(SIPAccount &account, RegistrationState state)
{
account.stopKeepAliveTimer();
account.setRegistrationState(state);
account.setRegister(false);
SIPVoIPLink::instance()->sipTransport.shutdownSipTransport(account);
}
void registration_cb(pjsip_regc_cbparam *param)
{
if (param == NULL) {
ERROR("registration callback parameter is NULL");
return;
}
SIPAccount *account = static_cast<SIPAccount *>(param->token);
if (account == NULL) {
ERROR("account doesn't exist in registration callback");
return;
}
if (account->isContactUpdateEnabled())
update_contact_header(param, account);
const pj_str_t *description = pjsip_get_status_text(param->code);
const std::string accountID = account->getAccountID();
if (param->code && description) {
std::string state(description->ptr, description->slen);
Manager::instance().getDbusManager()->getCallManager()->registrationStateChanged(accountID, state, param->code);
std::pair<int, std::string> details(param->code, state);
// TODO: there id a race condition for this ressource when closing the application
account->setRegistrationStateDetailed(details);
account->setRegistrationExpire(param->expiration);
}
#define FAILURE_MESSAGE() ERROR("Could not register account %s with error %d", accountID.c_str(), param->code)
if (param->status != PJ_SUCCESS) {
FAILURE_MESSAGE();
processRegistrationError(*account, ErrorAuth);
return;
}
if (param->code < 0 || param->code >= 300) {
switch (param->code) {
case PJSIP_SC_MULTIPLE_CHOICES: // 300
case PJSIP_SC_MOVED_PERMANENTLY: // 301
case PJSIP_SC_MOVED_TEMPORARILY: // 302
case PJSIP_SC_USE_PROXY: // 305
case PJSIP_SC_ALTERNATIVE_SERVICE: // 380
FAILURE_MESSAGE();
processRegistrationError(*account, Error);
break;
case PJSIP_SC_SERVICE_UNAVAILABLE: // 503
FAILURE_MESSAGE();
processRegistrationError(*account, ErrorHost);
break;
case PJSIP_SC_UNAUTHORIZED: // 401
// Automatically answered by PJSIP
account->registerVoIPLink();
break;
case PJSIP_SC_FORBIDDEN: // 403
case PJSIP_SC_NOT_FOUND: // 404
FAILURE_MESSAGE();
processRegistrationError(*account, ErrorAuth);
break;
case PJSIP_SC_REQUEST_TIMEOUT: // 408
FAILURE_MESSAGE();
processRegistrationError(*account, ErrorHost);
break;
case PJSIP_SC_INTERVAL_TOO_BRIEF: // 423
// Expiration Interval Too Brief
account->doubleRegistrationExpire();
account->registerVoIPLink();
account->setRegister(false);
break;
case PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE: // 606
lookForReceivedParameter(*param, *account);
account->setRegistrationState(ErrorNotAcceptable);
account->registerVoIPLink();
break;
default:
FAILURE_MESSAGE();
processRegistrationError(*account, Error);
break;
}
} else {
lookForReceivedParameter(*param, *account);
if (account->isRegistered())
account->setRegistrationState(Registered);
else {
account->setRegistrationState(Unregistered);
SIPVoIPLink::instance()->sipTransport.shutdownSipTransport(*account);
}
}
#undef FAILURE_MESSAGE
}
void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
{
SIPCall *currentCall = static_cast<SIPCall *>(inv->mod_data[mod_ua_.id]);
if (currentCall == NULL)
return;
static const pj_str_t str_refer_to = { (char*) "Refer-To", 8};
pjsip_generic_string_hdr *refer_to = static_cast<pjsip_generic_string_hdr*>
(pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL));
if (!refer_to) {
pjsip_dlg_respond(inv->dlg, rdata, 400, NULL, NULL, NULL);
return;
}
SIPVoIPLink::instance()->newOutgoingCall(Manager::instance().getNewCallID(), std::string(refer_to->hvalue.ptr, refer_to->hvalue.slen));
Manager::instance().hangupCall(currentCall->getCallId());
}
void transfer_client_cb(pjsip_evsub *sub, pjsip_event *event)
{
switch (pjsip_evsub_get_state(sub)) {
case PJSIP_EVSUB_STATE_ACCEPTED:
if (!event)
return;
pj_assert(event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
break;
case PJSIP_EVSUB_STATE_TERMINATED:
pjsip_evsub_set_mod_data(sub, mod_ua_.id, NULL);
break;
case PJSIP_EVSUB_STATE_ACTIVE: {
SIPVoIPLink *link = static_cast<SIPVoIPLink *>(pjsip_evsub_get_mod_data(sub, mod_ua_.id));
if (!link or !event)
return;
pjsip_rx_data* r_data = event->body.rx_msg.rdata;
if (!r_data)
return;
std::string request(pjsip_rx_data_get_info(r_data));
pjsip_status_line status_line = { 500, *pjsip_get_status_text(500) };
if (!r_data->msg_info.msg)
return;
if (r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD and
request.find("NOTIFY") != std::string::npos) {
pjsip_msg_body *body = r_data->msg_info.msg->body;
if (!body)
return;
if (pj_stricmp2(&body->content_type.type, "message") or
pj_stricmp2(&body->content_type.subtype, "sipfrag"))
return;
if (pjsip_parse_status_line((char*) body->data, body->len, &status_line) != PJ_SUCCESS)
return;
}
if (r_data->msg_info.cid)
return;
std::string transferID(r_data->msg_info.cid->id.ptr, r_data->msg_info.cid->id.slen);
SIPCall *call = dynamic_cast<SIPCall *>(link->getCall(transferCallID[transferID]));
if (!call)
return;
if (status_line.code / 100 == 2) {
pjsip_tx_data *tdata;
if (!call->inv)
return;
if (pjsip_inv_end_session(call->inv, PJSIP_SC_GONE, NULL, &tdata) == PJ_SUCCESS)
pjsip_inv_send_msg(call->inv, tdata);
Manager::instance().hangupCall(call->getCallId());
pjsip_evsub_set_mod_data(sub, mod_ua_.id, NULL);
}
break;
}
default:
break;
}
}
namespace {
unsigned int getRandomPort()
{
return ((rand() % 27250) + 5250) * 2;
}
}
void setCallMediaLocal(SIPCall* call, const std::string &localIP)
{
std::string account_id(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
if (!account)
return;
const unsigned int callLocalAudioPort = getRandomPort();
const unsigned int callLocalExternAudioPort = account->isStunEnabled()
? account->getStunPort()
: callLocalAudioPort;
call->setLocalIp(localIP);
call->setLocalAudioPort(callLocalAudioPort);
call->getLocalSDP()->setLocalPublishedAudioPort(callLocalExternAudioPort);
#ifdef SFL_VIDEO
unsigned int callLocalVideoPort = 0;
do
callLocalVideoPort = getRandomPort();
while (callLocalAudioPort == callLocalVideoPort);
call->setLocalVideoPort(callLocalVideoPort);
call->getLocalSDP()->setLocalPublishedVideoPort(callLocalVideoPort);
#endif
}
} // end anonymous namespace
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