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/*
Copyright (C) 2005-2009 Michel de Boer <michel@twinklephone.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 2 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, see <https://www.gnu.org/licenses/>.
*/
// Session description of an established session.
// A session is the media part of a dialog.
#ifndef _SESSION_H
#define _SESSION_H
#include <list>
#include <map>
#include <string>
#include "dialog.h"
#include "user.h"
#include "sdp/sdp.h"
#include "parser/sip_message.h"
#include "audio/audio_codecs.h"
#include "audio/audio_session.h"
// Forward declarations
class t_dialog;
class t_line;
using namespace std;
class t_session {
private:
// The owning dialog
t_dialog *dialog;
// User profile of user for which the session is created.
// This is a pointer to the user_config owned by a phone user.
// So this pointer should never be deleted.
t_user *user_config;
// Copy of host needed for call-retrieve after call-hold
string retrieve_host;
// Audio RTP session
t_audio_session *audio_rtp_session;
// Indicates if session is put on-hold, i.e. no RTP should be sent
// or received for this session.
bool is_on_hold;
// Indicates if a session is killed, i.e. RTP will never be
// sent or received anymore.
bool is_killed;
// Mapping from audio codecs to RTP payload numbers for receiving
// and sending directions.
map<t_audio_codec, unsigned short> recv_ac2payload;
map<t_audio_codec, unsigned short> send_ac2payload;
// Mapping from RTP payload numbers to audio codecs for receiving
// and sending directions.
map<unsigned short, t_audio_codec> recv_payload2ac;
map<unsigned short, t_audio_codec> send_payload2ac;
// Set the list of received codecs from the SDP.
// Create the send_ac2paylaod and send_payload2ac mappings.
void set_recvd_codecs(t_sdp *sdp);
// Returns if this session is part of a 3-way conference
bool is_3way(void) const;
// Returns the peer session of a 3-way conference
t_session *get_peer_3way (void) const;
public:
// Audio session information
// Near end information
string src_sdp_version;
string src_sdp_id;
string receive_host; // RTP receive host address
unsigned short receive_port; // RTP receive port
// Far end information
string dst_sdp_version;
string dst_sdp_id;
string dst_rtp_host;
unsigned short dst_rtp_port;
bool dst_zrtp_support;
// Direction of the audio stream from this phone's point of view
t_sdp_media_direction direction;
list<t_audio_codec> offer_codecs; // codecs to offer in outgoing INVITE
list<t_audio_codec> recvd_codecs; // codecs received from far-end
t_audio_codec use_codec; // codec to be used
unsigned short ptime; // payload size (ms)
unsigned short ilbc_mode; // 20 or 30 ms
bool recvd_offer; // offer received?
bool recvd_answer; // answer received?
bool sent_offer; // offer sent?
unsigned short recv_dtmf_pt; // payload type for DTMF receiving
unsigned short send_dtmf_pt; // payload type for DTMF sending
t_sdp recvd_sdp_offer;
t_session(t_dialog *_dialog, string _receive_host,
unsigned short _receive_port);
// The destructor will destroy the RTP session and stop the
// RTP streams
~t_session();
/** @name Clone a new session from an existing session. */
//@{
/** @note copies of a session do not copy the audio RTP session! */
/**
* Create a session based on an existing session, i.e.
* same receive user and host. The source SDP version of the
* new session will be increased by 1.
* @return The new session.
*/
t_session *create_new_version(void) const;
/**
* Create a copy of the session. The destination paramters
* and recvd/offer and answer are erased in the copy.
* The source SDP version of the new session will be increased by 1.
* @return The new session.
*/
t_session *create_clean_copy(void) const;
/**
* Create a session for call-hold.
* @return The call-hold session.
*/
t_session *create_call_hold(void) const;
/**
* Create a session for call-retrieve.
* @return The call-retrieve session.
*/
t_session *create_call_retrieve(void) const;
//@}
// Process incoming SDP offer. Return false if SDP is not
// supported. If SDP is supported then use_codec will be
// set to the first codec in the received offer that is
// supported by this phone, i.e. this is the codec that should
// be put in the answer.
bool process_sdp_offer(t_sdp *sdp, int &warn_code, string &warn_text);
// Process incoming SDP answer. Return false if SDP is not
// supported. It is expected that the answer contains 1 codec
// only. If more codecs are answered, then only the first supported
// codec is considered.
bool process_sdp_answer(t_sdp *sdp, int &warn_code, string &warn_text);
// Create an SDP offer body for a SIP message
void create_sdp_offer(t_sip_message *m, const string &user);
// Create an SDP answer body for a SIP message
void create_sdp_answer(t_sip_message *m, const string &user);
// Start/stop the RTP streams
// When a session is on-hold then start_rtp simply returns.
void start_rtp(void);
void stop_rtp(void);
// Kill RTP streams. The difference with stopping an RTP stream
// is that it cannot be started after being killed.
void kill_rtp(void);
t_audio_session *get_audio_session(void) const;
void set_audio_session(t_audio_session *as);
// Check if two session are equal wrt the audio parameters
bool equal_audio(const t_session &s) const;
// Send DTMF digit
void send_dtmf(char digit, bool inband);
// Get the line that belongs to this session
t_line *get_line(void) const;
// Transfer ownership of this session to a new dialog
void set_owner(t_dialog *d);
// Hold/un-hold a session
// These methods only toggle the hold indicator. If you hold
// a session, you must make sure that any running RTP is stopped.
// If you unhold a session you have to call start_rtp to start the
// RTP.
void hold(void);
void unhold(void);
// Check if RTP session is acitve
bool is_rtp_active(void) const;
};
#endif
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