sipvoiplink.cpp 57.3 KB
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/*
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 *  Copyright (C) 2004, 2005, 2006, 2008, 2009, 2010 Savoir-Faire Linux Inc.
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 *  Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
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 *  Author: Yun Liu <yun.liu@savoirfairelinux.com>
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 *  Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
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 *  Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
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 *
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 *  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
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 *  the Free Software Foundation; either version 3 of the License, or
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 *  (at your option) any later version.
 *
 *  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.
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 *
 *  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.
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 */
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#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

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#include "sip_utils.h"

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#include "sipvoiplink.h"
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#include "manager.h"
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#include "logger.h"
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#include "sip/sdp.h"
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#include "sipcall.h"
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#include "sipaccount.h"
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#include "eventthread.h"
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#include "sdes_negotiator.h"
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#include "dbus/dbusmanager.h"
#include "dbus/callmanager.h"
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#include "dbus/configurationmanager.h"
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#include "im/instant_messaging.h"
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#include "audio/audiolayer.h"

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#include "pjsip/sip_endpoint.h"
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#include "pjsip/sip_transport_tls.h"
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#include "pjsip/sip_uri.h"
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#include "pjnath.h"
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#include <netinet/in.h>
#include <arpa/nameser.h>
#include <resolv.h>
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#include <istream>
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#include <utility> // for std::pair
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#include <map>

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using namespace sfl;

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SIPVoIPLink *SIPVoIPLink::instance_ = 0;
bool SIPVoIPLink::destroyed_ = false;

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namespace {
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/** A map to retreive SFLphone internal call id
 *  Given a SIP call ID (usefull for transaction sucha as transfer)*/
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static std::map<std::string, std::string> transferCallID;
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/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
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/**
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 * Set audio (SDP) configuration for a call
 * localport, localip, localexternalport
 * @param call a SIPCall valid pointer
 */
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void setCallMediaLocal(SIPCall* call, const std::string &localIP);
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static pj_caching_pool pool_cache, *cp_ = &pool_cache;
static pj_pool_t *pool_;
static pjsip_endpoint *endpt_;
static pjsip_module mod_ua_;
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static pj_thread_t *thread_;
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void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status);
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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);
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pj_bool_t transaction_response_cb(pjsip_rx_data *rdata) ;
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void transfer_client_cb(pjsip_evsub *sub, pjsip_event *event);
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/**
 * 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 *);

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/**
 * Helper function to process refer function on call transfer
 */
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void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata);
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/*
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pjsip_route_hdr *createRouteSet(const std::string &route, pj_pool_t *hdr_pool)
{
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    int port = 0;
    std::string host;
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    size_t found = route.find(":");
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    if (found != std::string::npos) {
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        host = route.substr(0, found);
        port = atoi(route.substr(found + 1, route.length()).c_str());
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    } else
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        host = route;

    pjsip_route_hdr *route_set = pjsip_route_hdr_create(hdr_pool);
    pjsip_route_hdr *routing = pjsip_route_hdr_create(hdr_pool);
    pjsip_sip_uri *url = pjsip_sip_uri_create(hdr_pool, 0);
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    routing->name_addr.uri = (pjsip_uri*) url;
    pj_strdup2(hdr_pool, &url->host, host.c_str());
    url->port = port;

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    pj_list_push_back(route_set, pjsip_hdr_clone(hdr_pool, routing));
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    return route_set;
}
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*/
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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)); \
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} while (0)
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#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);
}

pj_bool_t transaction_response_cb(pjsip_rx_data *rdata)
{
    pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);

    if (!dlg)
        return PJ_SUCCESS;

    pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);

    if (!tsx or tsx->method.id != PJSIP_INVITE_METHOD)
        return PJ_SUCCESS;

    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;
        pjsip_dlg_create_request(dlg, &pjsip_ack_method, rdata->msg_info.cseq->cseq, &tdata);
        pjsip_dlg_send_request(dlg, tdata, -1, NULL);
    }

    return PJ_SUCCESS;
}

pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
{
    pjsip_method *method = &rdata->msg_info.msg->line.req.method;

    if (method->id == PJSIP_ACK_METHOD && pjsip_rdata_get_dlg(rdata))
        return 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);
    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));

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    std::string displayName(sip_utils::parseDisplayName(rdata->msg_info.msg_buf));
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    if (method->id == PJSIP_OTHER_METHOD) {
        pj_str_t *str = &method->name;
        std::string request(str->ptr, str->slen);

        if (request.find("NOTIFY") != (size_t)-1) {
            int voicemail;

            if (sscanf((const char*)rdata->msg_info.msg->body->data, "Voice-Message: %d/", &voicemail) == 1 && voicemail != 0)
                Manager::instance().startVoiceMessageNotification(account_id, voicemail);
        }

        pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_OK, NULL, NULL, NULL);

        return true;
    } else if (method->id == PJSIP_OPTIONS_METHOD) {
        handleIncomingOptions(rdata);
        return 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 true;
    }

    SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));

    pjmedia_sdp_session *r_sdp;
    pjsip_msg_body *body = rdata->msg_info.msg->body;

    if (!body || pjmedia_sdp_parse(rdata->tp_info.pool, (char*) body->data, body->len, &r_sdp) != PJ_SUCCESS)
        r_sdp = NULL;

    if (account->getActiveCodecs().empty()) {
        pjsip_endpt_respond_stateless(endpt_, rdata,
                                      PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL,
                                      NULL);
        return false;
    }

    // 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 true;
    }

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    Manager::instance().hookPreference.runHook(rdata->msg_info.msg);
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    SIPCall* call = new SIPCall(Manager::instance().getNewCallID(), Call::INCOMING, cp_);
    Manager::instance().associateCallToAccount(call->getCallId(), account_id);

    // May use the published address as well
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    std::string addrToUse = SipTransport::getInterfaceAddrFromName(account->getLocalInterface());
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    std::string addrSdp = account->isStunEnabled()
                          ? account->getPublishedAddress()
                          : addrToUse;

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    pjsip_tpselector *tp = SIPVoIPLink::instance()->sipTransport.initTransportSelector(account->transport_, call->getMemoryPool());
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    if (addrToUse == "0.0.0.0")
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        addrToUse = SipTransport::getSIPLocalIP();
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    if (addrSdp == "0.0.0.0")
        addrSdp = addrToUse;

    char tmp[PJSIP_MAX_URL_SIZE];
    int length = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, sip_from_uri, tmp, PJSIP_MAX_URL_SIZE);
    std::string peerNumber(tmp, length);
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    sip_utils::stripSipUriPrefix(peerNumber);
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    call->setConnectionState(Call::PROGRESSING);
    call->setPeerNumber(peerNumber);
    call->setDisplayName(displayName);
    call->initRecFilename(peerNumber);

    setCallMediaLocal(call, addrToUse);

    call->getLocalSDP()->setLocalIP(addrSdp);

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    call->getAudioRtp().initConfig();
    call->getAudioRtp().initSession();
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    if (rdata->msg_info.msg->body) {
        char sdpbuffer[1000];
        int len = rdata->msg_info.msg->body->print_body(rdata->msg_info.msg->body, sdpbuffer, sizeof sdpbuffer);

        if (len == -1) // error
            len = 0;

        std::string sdpoffer(sdpbuffer, 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)));

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            std::vector<sfl::CryptoSuiteDefinition> localCapabilities;
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            for (int i = 0; i < 3; i++)
                localCapabilities.push_back(sfl::CryptoSuites[i]);

            sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);

            if (sdesnego.negotiate()) {
                call->getAudioRtp().setRemoteCryptoInfo(sdesnego);
                call->getAudioRtp().initLocalCryptoInfo();
            }
        }
    }

    call->getLocalSDP()->receiveOffer(r_sdp, account->getActiveCodecs());

    sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec(PAYLOAD_CODEC_ULAW);
    call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));

    pjsip_dialog* dialog;

    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 false;
    }

    pjsip_inv_create_uas(dialog, rdata, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv);

    PJ_ASSERT_RETURN(pjsip_dlg_set_transport(dialog, tp) == PJ_SUCCESS, 1);

    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.");
        pjsip_endpt_respond_stateless(endpt_, rdata, 500 /* internal server error */, NULL, NULL, NULL);
    }

    // Check if call has been transfered
    pjsip_tx_data *tdata;

    if (replaced_dlg) { // If Replace header present
        // 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
        PJ_ASSERT_RETURN(pjsip_inv_initial_answer(call->inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata) == PJ_SUCCESS, 1);
        PJ_ASSERT_RETURN(pjsip_inv_send_msg(call->inv, tdata) == PJ_SUCCESS, 1);

        call->setConnectionState(Call::RINGING);

        Manager::instance().incomingCall(call, account_id);
        Manager::instance().getAccountLink(account_id)->addCall(call);
    }

    return true;
}
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} // end anonymous namespace

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/*************************************************************************************************/

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SIPVoIPLink::SIPVoIPLink() : sipTransport(endpt_, cp_, pool_), evThread_(this)
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{
#define TRY(ret) do { \
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    if (ret != PJ_SUCCESS) \
    throw VoipLinkException(#ret " failed"); \
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} while (0)
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    srand(time(NULL)); // to get random number for RANDOM_PORT
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    TRY(pj_init());
    TRY(pjlib_util_init());
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    // From 0 (min) to 6 (max)
    pj_log_set_level(Logger::getDebugMode() ? 6 : 0);
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    TRY(pjnath_init());
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    pj_caching_pool_init(cp_, &pj_pool_factory_default_policy, 0);
    pool_ = pj_pool_create(&cp_->factory, "sflphone", 4000, 4000, NULL);
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    if (!pool_)
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        throw VoipLinkException("UserAgent: Could not initialize memory pool");
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    TRY(pjsip_endpt_create(&cp_->factory, pj_gethostname()->ptr, &endpt_));
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    sipTransport.setEndpoint(endpt_);
    sipTransport.setCachingPool(cp_);
    sipTransport.setPool(pool_);

    if (SipTransport::getSIPLocalIP().empty())
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        throw VoipLinkException("UserAgent: Unable to determine network capabilities");

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    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_));
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    // Initialize and register sflphone module
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    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_));
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    TRY(pjsip_evsub_init_module(endpt_));
    TRY(pjsip_xfer_init_module(endpt_));
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    static const pjsip_inv_callback inv_cb = {
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        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,
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    };
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    TRY(pjsip_inv_usage_init(endpt_, &inv_cb));
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    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}
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    pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
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    static const pj_str_t text_plain = { (char*) "text/plain", 10 };
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    pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, NULL, 1, &text_plain);
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    static const pj_str_t accepted = { (char*) "application/sdp", 15 };
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    pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, NULL, 1, &accepted);
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    DEBUG("UserAgent: pjsip version %s for %s initialized", pj_get_version(), PJ_OS_NAME);
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    TRY(pjsip_replaces_init_module(endpt_));
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#undef TRY
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    handlingEvents_ = true;
    evThread_.start();
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}
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SIPVoIPLink::~SIPVoIPLink()
{
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    handlingEvents_ = false;
    if (thread_) {
        pj_thread_join(thread_);
        pj_thread_destroy(thread_);
        DEBUG("PJ thread destroy finished");
        thread_ = 0;
    }
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    const pj_time_val tv = {0, 10};
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    pjsip_endpt_handle_events(endpt_, &tv);
    pjsip_endpt_destroy(endpt_);
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    pj_pool_release(pool_);
    pj_caching_pool_destroy(cp_);
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    pj_shutdown();
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}

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SIPVoIPLink* SIPVoIPLink::instance()
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{
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    assert(!destroyed_);
    if (!instance_)
        instance_ = new SIPVoIPLink;
    return instance_;
}

void SIPVoIPLink::destroy()
{
    delete instance_;
    destroyed_ = true;
    instance_ = 0;
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}

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// Called from EventThread::run (not main thread)
bool SIPVoIPLink::getEvent()
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{
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    static pj_thread_desc desc;
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    // We have to register the external thread so it could access the pjsip frameworks
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    if (!pj_thread_is_registered()) {
        DEBUG("%s: Registering thread", __PRETTY_FUNCTION__);
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        pj_thread_register(NULL, desc, &thread_);
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    }
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    static const pj_time_val timeout = {0, 10};
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    pjsip_endpt_handle_events(endpt_, &timeout);
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    return handlingEvents_;
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}
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void SIPVoIPLink::sendRegister(Account *a)
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{
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    SIPAccount *account = dynamic_cast<SIPAccount*>(a);
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    sipTransport.createSipTransport(account);
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    account->setRegister(true);
    account->setRegistrationState(Trying);
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    pjsip_regc *regc = account->getRegistrationInfo();
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    if (pjsip_regc_create(endpt_, (void *) account, &registration_cb, &regc) != PJ_SUCCESS)
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        throw VoipLinkException("UserAgent: Unable to create regc structure.");

    std::string srvUri(account->getServerUri());
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    // std::string address, port;
    // findLocalAddressFromUri(srvUri, account->transport_, address, port);
    pj_str_t pjSrv = pj_str((char*) srvUri.c_str());
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    // Generate the FROM header
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    std::string from(account->getFromUri());
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    pj_str_t pjFrom = pj_str((char*) from.c_str());
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    // Get the contact header for this account
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    std::string contact(account->getContactHeader());
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    pj_str_t pjContact = pj_str((char*) contact.c_str());
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    if (pjsip_regc_init(regc, &pjSrv, &pjFrom, &pjFrom, 1, &pjContact, account->getRegistrationExpire()) != PJ_SUCCESS)
        throw VoipLinkException("Unable to initialize account registration structure");
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    if (!account->getServiceRoute().empty())
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        pjsip_regc_set_route_set(regc, sip_utils::createRouteSet(account->getServiceRoute(), pool_));
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    pjsip_regc_set_credentials(regc, account->getCredentialCount(), account->getCredInfo());
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    pjsip_hdr hdr_list;
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    pj_list_init(&hdr_list);
    std::string useragent(account->getUserAgentName());
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    pj_str_t pJuseragent = pj_str((char*) useragent.c_str());
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    const pj_str_t STR_USER_AGENT = { (char*) "User-Agent", 10 };
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    pjsip_generic_string_hdr *h = pjsip_generic_string_hdr_create(pool_, &STR_USER_AGENT, &pJuseragent);
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    pj_list_push_back(&hdr_list, (pjsip_hdr*) h);
    pjsip_regc_add_headers(regc, &hdr_list);
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    pjsip_tx_data *tdata;
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    if (pjsip_regc_register(regc, PJ_TRUE, &tdata) != PJ_SUCCESS)
        throw VoipLinkException("Unable to initialize transaction data for account registration");
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    if (pjsip_regc_set_transport(regc, sipTransport.initTransportSelector(account->transport_, pool_)) != PJ_SUCCESS)
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        throw VoipLinkException("Unable to set transport");
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    // decrease transport's ref count, counter incrementation is managed when acquiring transport
    pjsip_transport_dec_ref(account->transport_);
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    // 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");
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    // 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);
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    // start the periodic registration request based on Expire header
    // account determines itself if a keep alive is required
    account->startKeepAliveTimer();
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}

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void SIPVoIPLink::sendUnregister(Account *a)
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{
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    SIPAccount *account = dynamic_cast<SIPAccount *>(a);
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    // This may occurs if account failed to register and is in state INVALID
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    if (!account->isRegistered()) {
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        account->setRegistrationState(Unregistered);
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        return;
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    }

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    // Make sure to cancel any ongoing timers before unregister
    account->stopKeepAliveTimer();

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    pjsip_regc *regc = account->getRegistrationInfo();
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    if (!regc)
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        throw VoipLinkException("Registration structure is NULL");
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    pjsip_tx_data *tdata = NULL;
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    if (pjsip_regc_unregister(regc, &tdata) != PJ_SUCCESS)
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        throw VoipLinkException("Unable to unregister sip account");
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    if (pjsip_regc_send(regc, tdata) != PJ_SUCCESS)
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        throw VoipLinkException("Unable to send request to unregister sip account");
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    account->setRegister(false);
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}

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void SIPVoIPLink::registerKeepAliveTimer(pj_timer_entry &timer, pj_time_val &delay)
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{
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    if (timer.id == -1)
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        WARN("UserAgent: Timer already scheduled");

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    switch (pjsip_endpt_schedule_timer(endpt_, &timer, &delay)) {
        case PJ_SUCCESS:
            break;
        default:
            ERROR("UserAgent: Could not schedule new timer in pjsip endpoint");
            /* fallthrough */
        case PJ_EINVAL:
            ERROR("UserAgent: Invalid timer or delay entry");
            break;
        case PJ_EINVALIDOP:
            ERROR("Invalid timer entry, maybe already scheduled");
            break;
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    }
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}

void SIPVoIPLink::cancelKeepAliveTimer(pj_timer_entry& timer)
{
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    pjsip_endpt_cancel_timer(endpt_, &timer);
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}

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Call *SIPVoIPLink::newOutgoingCall(const std::string& id, const std::string& toUrl)
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{
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    SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(Manager::instance().getAccountFromCall(id)));

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    if (account == NULL) // TODO: We should investigate how we could get rid of this error and create a IP2IP call instead
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        throw VoipLinkException("Could not get account for this call");
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    SIPCall* call = new SIPCall(id, Call::OUTGOING, cp_);
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    // If toUri is not a well formatted sip URI, use account information to process it
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    std::string toUri;
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    if (toUrl.find("sip:") != std::string::npos or
        toUrl.find("sips:") != std::string::npos)
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        toUri = toUrl;
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    else
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        toUri = account->getToUri(toUrl);
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    call->setPeerNumber(toUri);
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    std::string localAddr(SipTransport::getInterfaceAddrFromName(account->getLocalInterface()));
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    if (localAddr == "0.0.0.0")
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        localAddr = SipTransport::getSIPLocalIP();
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    setCallMediaLocal(call, localAddr);
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    // May use the published address as well
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    std::string addrSdp = account->isStunEnabled() ?
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    account->getPublishedAddress() :
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    SipTransport::getInterfaceAddrFromName(account->getLocalInterface());
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    if (addrSdp == "0.0.0.0")
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        addrSdp = SipTransport::getSIPLocalIP();
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    // 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
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    sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec(PAYLOAD_CODEC_ULAW);

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    if (audiocodec == NULL) {
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        delete call;
        throw VoipLinkException("Could not instantiate codec for early media");
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    }
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    try {
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        call->getAudioRtp().initConfig();
        call->getAudioRtp().initSession();
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        call->getAudioRtp().initLocalCryptoInfo();
        call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
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    } catch (...) {
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        delete call;
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        throw VoipLinkException("Could not start rtp session for early media");
    }
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    call->initRecFilename(toUrl);
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    call->getLocalSDP()->setLocalIP(addrSdp);
    call->getLocalSDP()->createOffer(account->getActiveCodecs());
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    if (!SIPStartCall(call)) {
        delete call;
        throw VoipLinkException("Could not send outgoing INVITE request for new call");
    }
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    return call;
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}
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void
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SIPVoIPLink::answer(Call *call)
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{
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    if (!call)
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        return;
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    call->answer();
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}

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void
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SIPVoIPLink::hangup(const std::string& id)
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{
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    SIPCall* call = getSIPCall(id);
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    std::string account_id(Manager::instance().getAccountFromCall(id));
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    SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));

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    if (account == NULL)
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        throw VoipLinkException("Could not find account for this call");
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    pjsip_inv_session *inv = call->inv;
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    if (inv == NULL)
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        throw VoipLinkException("No invite session for this call");
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    // Looks for sip routes
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    if (not account->getServiceRoute().empty()) {
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        pjsip_route_hdr *route_set = sip_utils::createRouteSet(account->getServiceRoute(), inv->pool);
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        pjsip_dlg_set_route_set(inv->dlg, route_set);
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    }
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    pjsip_tx_data *tdata = NULL;
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    // User hangup current call. Notify peer
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    if (pjsip_inv_end_session(inv, 404, NULL, &tdata) != PJ_SUCCESS || !tdata)
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        return;
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    if (pjsip_inv_send_msg(inv, tdata) != PJ_SUCCESS)
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        return;
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    // Make sure user data is NULL in callbacks
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    inv->mod_data[mod_ua_.id] = NULL;
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    if (Manager::instance().isCurrentCall(id))
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        call->getAudioRtp().stop();
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    removeCall(id);
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}

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void
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SIPVoIPLink::peerHungup(const std::string& id)
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{
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    SIPCall* call = getSIPCall(id);
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    // User hangup current call. Notify peer
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    pjsip_tx_data *tdata = NULL;
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    if (pjsip_inv_end_session(call->inv, 404, NULL, &tdata) != PJ_SUCCESS || !tdata)
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        return;
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    if (pjsip_inv_send_msg(call->inv, tdata) != PJ_SUCCESS)
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        return;
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    // Make sure user data is NULL in callbacks
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    call->inv->mod_data[mod_ua_.id ] = NULL;
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    if (Manager::instance().isCurrentCall(id))
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        call->getAudioRtp().stop();
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    removeCall(id);
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}

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void
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SIPVoIPLink::onhold(const std::string& id)
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{
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    SIPCall *call = getSIPCall(id);
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    call->setState(Call::HOLD);
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    call->getAudioRtp().stop();
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    Sdp *sdpSession = call->getLocalSDP();
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    if (!sdpSession)
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        throw VoipLinkException("Could not find sdp session");
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    sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
    sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
    sdpSession->addAttributeToLocalAudioMedia("sendonly");

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    SIPSessionReinvite(call);
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}

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void
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SIPVoIPLink::offhold(const std::string& id)
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{
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    SIPCall *call = getSIPCall(id);

    Sdp *sdpSession = call->getLocalSDP();
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    if (sdpSession == NULL)
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        throw VoipLinkException("Could not find sdp session");
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    try {
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        int pl = PAYLOAD_CODEC_ULAW;
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        sfl::Codec *sessionMedia = sdpSession->getSessionMedia();
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        if (sessionMedia)
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            pl = sessionMedia->getPayloadType();
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