sipcall.h

/*
 *  Copyright (C) 2004-2017 Savoir-faire Linux Inc.
 *
 *  Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
 *  Author: Yan Morin <yan.morin@savoirfairelinux.com>
 *  Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
 *  Author: Guillaume Roguez <guillaume.roguez@savoirfairelinux.com>
 *
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; either version 3 of the License, or
 *  (at your option) any later version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *  Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA.
 */

#ifndef __SIPCALL_H__
#define __SIPCALL_H__

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include "call.h"
#include "sip_utils.h"

#ifdef RING_VIDEO
#include "media/video/video_rtp_session.h"
#endif

#include "noncopyable.h"

#include "pjsip/sip_config.h"

#include <memory>

struct pjsip_evsub;
struct pjsip_inv_session;
struct pjmedia_sdp_session;

namespace ring {

class Sdp;
class SIPAccountBase;
class SipTransport;
class AudioRtpSession;

namespace upnp {
class Controller;
}

/**
 * @file sipcall.h
 * @brief SIPCall are SIP implementation of a normal Call
 */
class SIPCall : public Call
{
    public:
        static const char* const LINK_TYPE;

    protected:
        /**
         * Constructor (protected)
         * @param id	The call identifier
         * @param type  The type of the call. Could be Incoming
         *						 Outgoing
         */
        SIPCall(SIPAccountBase& account, const std::string& id, Call::CallType type);

    public:
        /**
         * Destructor
         */
        ~SIPCall();

        /**
         * Return the SDP's manager of this call
         */
        Sdp& getSDP() {
            return *sdp_;
        }

        const char* getLinkType() const override {
            return LINK_TYPE;
        }

        /**
         * Returns a pointer to the AudioRtpSession object
         */
        AudioRtpSession& getAVFormatRTP() const {
            return *avformatrtp_;
        }

#ifdef RING_VIDEO
        /**
         * Returns a pointer to the VideoRtp object
         */
        video::VideoRtpSession& getVideoRtp () {
            return *videortp_;
        }
#endif

        /**
         * The invite session to be reused in case of transfer
         */
        struct InvSessionDeleter {
                void operator()(pjsip_inv_session*) const noexcept;
        };

        std::unique_ptr<pjsip_inv_session, InvSessionDeleter> inv;

        void setSecure(bool sec);

        bool isSecure() const {
            return srtpEnabled_;
        }

        void setCallMediaLocal(const pj_sockaddr& localIP);

        void generateMediaPorts();

        void setContactHeader(pj_str_t *contact);

        void setTransport(const std::shared_ptr<SipTransport>& t);

        inline SipTransport* getTransport() {
            return transport_.get();
        }

        void sendSIPInfo(const char *const body, const char *const subtype);

        void answer() override;

        void hangup(int reason) override;

        void refuse() override;

        void transfer(const std::string& to) override;

        bool attendedTransfer(const std::string& to) override;

        bool onhold() override;

        bool offhold() override;

        void switchInput(const std::string& resource) override;

        void peerHungup() override;

        void carryingDTMFdigits(char code) override;

        void sendTextMessage(const std::map<std::string, std::string>& messages,
                             const std::string& from) override;

        void removeCall() override;

        SIPAccountBase& getSIPAccount() const;

        void updateSDPFromSTUN();

        /**
         * Tell the user that the call is ringing
         * @param
         */
        void onPeerRinging();

        /**
         * Tell the user that the call was answered
         * @param
         */
        void onAnswered();

        /**
         * To call in case of server/internal error
         * @param cause Optionnal error code
         */
        void onFailure(signed cause=0);

        /**
         * Peer close the connection
         * @param
         */
        void onClosed();

        void setupLocalSDPFromIce();

        bool startIce();

        void startAllMedia();

        void onMediaUpdate();

        void onReceiveOffer(const pjmedia_sdp_session *offer);

        void openPortsUPnP();

        void muteMedia(const std::string& mediaType, bool isMuted) override;

        void restartMediaSender() override;

        void restartMediaReceiver() override;

        bool useVideoCodec(const AccountVideoCodecInfo* codec) const override;

        virtual std::map<std::string, std::string> getDetails() const override;

        bool initIceTransport(bool master, unsigned channel_num=4) override;

        void terminateSipSession(int status);

        virtual void merge(const std::shared_ptr<Call>& scall) {
            merge(std::dynamic_pointer_cast<SIPCall>(scall));
        }
        virtual void merge(const std::shared_ptr<SIPCall>& scall);

        void setPeerRegistredName(const std::string& name) {
            peerRegistredName_ = name;
        }

    private:
        NON_COPYABLE(SIPCall);

        void waitForIceAndStartMedia();

        void stopAllMedia();

        /**
         * Transfer method used for both type of transfer
         */
        bool transferCommon(pj_str_t *dst);

        bool internalOffHold(const std::function<void()> &SDPUpdateFunc);

        int SIPSessionReinvite();

        std::vector<IceCandidate> getAllRemoteCandidates();

        std::unique_ptr<AudioRtpSession> avformatrtp_;

#ifdef RING_VIDEO
        /**
         * Video Rtp Session factory
         */
        std::unique_ptr<video::VideoRtpSession> videortp_;

        std::string videoInput_;
#endif

        bool srtpEnabled_ {false};

        /**
         * Hold the transport used for SIP communication.
         * Will be different from the account registration transport for
         * non-IP2IP calls.
         */
        std::shared_ptr<SipTransport> transport_ {};

        /**
         * The SDP session
         */
        std::unique_ptr<Sdp> sdp_;
        bool peerHolding_ {false};

        std::string peerRegistredName_ {};

        char contactBuffer_[PJSIP_MAX_URL_SIZE] {};
        pj_str_t contactHeader_ {contactBuffer_, 0};

        std::unique_ptr<ring::upnp::Controller> upnp_;
};

} // namespace ring

#endif // __SIPCALL_H__