Samples: Using SIP and Custom RTP/RTCP to Monitor Quality

This source is an example to demonstrate using SIP and RTP/RTCP framework to measure the network quality/impairment from the SIP call. This program can be used to make calls or to receive calls from other SIP endpoint (or other siprtp program), and to display the media quality statistics at the end of the call.

Note that the remote peer must support RTCP.

The layout of the program has been designed so that custom reporting can be generated instead of plain human readable text.

The source code of the file is pjsip-apps/src/samples/siprtp.c

Screenshots on WinXP:

/* 
 * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
 * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
 *
 * 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, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA 
 */
 
 
 
 
 
/* Usage */
static const char *USAGE = 
" PURPOSE:                                                                  \n"
"   This program establishes SIP INVITE session and media, and calculate    \n"
"   the media quality (packet lost, jitter, rtt, etc.). Unlike normal       \n"
"   pjmedia applications, this program bypasses all pjmedia stream          \n"
"   framework and transmit encoded RTP packets manually using own thread.   \n"
"\n"
" USAGE:\n"
"   siprtp [options]        => to start in server mode\n"
"   siprtp [options] URL    => to start in client mode\n"
"\n"
" Program options:\n"
"   --count=N,        -c    Set number of calls to create (default:1) \n"
"   --gap=N           -g    Set call gapping to N msec (default:0)\n"
"   --duration=SEC,   -d    Set maximum call duration (default:unlimited) \n"
"   --auto-quit,      -q    Quit when calls have been completed (default:no)\n"
"   --call-report     -R    Display report on call termination (default:yes)\n"
"\n"
" Address and ports options:\n"
"   --local-port=PORT,-p    Set local SIP port (default: 5060)\n"
"   --rtp-port=PORT,  -r    Set start of RTP port (default: 4000)\n"
"   --ip-addr=IP,     -i    Set local IP address to use (otherwise it will\n"
"                           try to determine local IP address from hostname)\n"
"\n"
" Logging Options:\n"
"   --log-level=N,    -l    Set log verbosity level (default=5)\n"
"   --app-log-level=N       Set app screen log verbosity (default=3)\n"
"   --log-file=FILE         Write log to file FILE\n"
"   --report-file=FILE      Write report to file FILE\n"
"\n"
/* Don't support this anymore, because codec is properly examined in
   pjmedia_session_info_from_sdp() function.
 
" Codec Options:\n"
"   --a-pt=PT               Set audio payload type to PT (default=0)\n"
"   --a-name=NAME           Set audio codec name to NAME (default=pcmu)\n"
"   --a-clock=RATE          Set audio codec rate to RATE Hz (default=8000Hz)\n"
"   --a-bitrate=BPS         Set audio codec bitrate to BPS (default=64000bps)\n"
"   --a-ptime=MS            Set audio frame time to MS msec (default=20ms)\n"
*/
;
 
 
/* Include all headers. */
#include <pjsip.h>
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pjsip_ua.h>
#include <pjsip_simple.h>
#include <pjlib-util.h>
#include <pjlib.h>
 
#include <stdlib.h>
 
/* Uncomment these to disable threads.
 * NOTE:
 *   when threading is disabled, siprtp won't transmit any
 *   RTP packets.
 */
/*
#undef PJ_HAS_THREADS
#define PJ_HAS_THREADS 0
*/
 
 
#if PJ_HAS_HIGH_RES_TIMER==0
#   error "High resolution timer is needed for this sample"
#endif
 
#define THIS_FILE       "siprtp.c"
#define MAX_CALLS       1024
#define RTP_START_PORT  4000
 
 
/* Codec descriptor: */
struct codec
{
    unsigned    pt;
    char*       name;
    unsigned    clock_rate;
    unsigned    bit_rate;
    unsigned    ptime;
    char*       description;
};
 
 
/* A bidirectional media stream created when the call is active. */
struct media_stream
{
    /* Static: */
    unsigned             call_index;        /* Call owner.              */
    unsigned             media_index;       /* Media index in call.     */
    pjmedia_transport   *transport;         /* To send/recv RTP/RTCP    */
 
    /* Active? */
    pj_bool_t            active;            /* Non-zero if is in call.  */
 
    /* Current stream info: */
    pjmedia_stream_info  si;                /* Current stream info.     */
 
    /* More info: */
    unsigned             clock_rate;        /* clock rate               */
    unsigned             samples_per_frame; /* samples per frame        */
    unsigned             bytes_per_frame;   /* frame size.              */
 
    /* RTP session: */
    pjmedia_rtp_session  out_sess;          /* outgoing RTP session     */
    pjmedia_rtp_session  in_sess;           /* incoming RTP session     */
 
    /* RTCP stats: */
    pjmedia_rtcp_session rtcp;              /* incoming RTCP session.   */
 
    /* Thread: */
    pj_bool_t            thread_quit_flag;  /* Stop media thread.       */
    pj_thread_t         *thread;            /* Media thread.            */
};
 
 
/* This is a call structure that is created when the application starts
 * and only destroyed when the application quits.
 */
struct call
{
    unsigned             index;
    pjsip_inv_session   *inv;
    unsigned             media_count;
    struct media_stream  media[1];
    pj_time_val          start_time;
    pj_time_val          response_time;
    pj_time_val          connect_time;
 
    pj_timer_entry       d_timer;           
};
 
 
/* Application's global variables */
static struct app
{
    unsigned             max_calls;
    unsigned             call_gap;
    pj_bool_t            call_report;
    unsigned             uac_calls;
    unsigned             duration;
    pj_bool_t            auto_quit;
    unsigned             thread_count;
    int                  sip_port;
    int                  rtp_start_port;
    pj_str_t             local_addr;
    pj_str_t             local_uri;
    pj_str_t             local_contact;
    
    int                  app_log_level;
    int                  log_level;
    char                *log_filename;
    char                *report_filename;
 
    struct codec         audio_codec;
 
    pj_str_t             uri_to_call;
 
    pj_caching_pool      cp;
    pj_pool_t           *pool;
 
    pjsip_endpoint      *sip_endpt;
    pj_bool_t            thread_quit;
    pj_thread_t         *sip_thread[1];
 
    pjmedia_endpt       *med_endpt;
    struct call          call[MAX_CALLS];
} app;
 
 
 
/*
 * Prototypes:
 */
 
/* Callback to be called when SDP negotiation is done in the call: */
static void call_on_media_update( pjsip_inv_session *inv,
                                  pj_status_t status);
 
/* Callback to be called when invite session's state has changed: */
static void call_on_state_changed( pjsip_inv_session *inv, 
                                   pjsip_event *e);
 
/* Callback to be called when dialog has forked: */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
 
/* Callback to be called to handle incoming requests outside dialogs: */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata );
 
/* Worker thread prototype */
static int sip_worker_thread(void *arg);
 
/* Create SDP for call */
static pj_status_t create_sdp( pj_pool_t *pool,
                               struct call *call,
                               pjmedia_sdp_session **p_sdp);
 
/* Hangup call */
static void hangup_call(unsigned index);
 
/* Destroy the call's media */
static void destroy_call_media(unsigned call_index);
 
/* Destroy media. */
static void destroy_media();
 
/* This callback is called by media transport on receipt of RTP packet. */
static void on_rx_rtp(void *user_data, void *pkt, pj_ssize_t size);
 
/* This callback is called by media transport on receipt of RTCP packet. */
static void on_rx_rtcp(void *user_data, void *pkt, pj_ssize_t size);
 
/* Display error */
static void app_perror(const char *sender, const char *title, 
                       pj_status_t status);
 
/* Print call */
static void print_call(int call_index);
 
 
/* This is a PJSIP module to be registered by application to handle
 * incoming requests outside any dialogs/transactions. The main purpose
 * here is to handle incoming INVITE request message, where we will
 * create a dialog and INVITE session for it.
 */
static pjsip_module mod_siprtp =
{
    NULL, NULL,                     /* prev, next.              */
    { "mod-siprtpapp", 13 },        /* Name.                    */
    -1,                             /* Id                       */
    PJSIP_MOD_PRIORITY_APPLICATION, /* Priority                 */
    NULL,                           /* load()                   */
    NULL,                           /* start()                  */
    NULL,                           /* stop()                   */
    NULL,                           /* unload()                 */
    &on_rx_request,                 /* on_rx_request()          */
    NULL,                           /* on_rx_response()         */
    NULL,                           /* on_tx_request.           */
    NULL,                           /* on_tx_response()         */
    NULL,                           /* on_tsx_state()           */
};
 
 
/* Codec constants */
struct codec audio_codecs[] = 
{
    { 0,  "PCMU", 8000, 64000, 20, "G.711 ULaw" },
    { 3,  "GSM",  8000, 13200, 20, "GSM" },
    { 4,  "G723", 8000, 6400,  30, "G.723.1" },
    { 8,  "PCMA", 8000, 64000, 20, "G.711 ALaw" },
    { 18, "G729", 8000, 8000,  20, "G.729" },
};
 
 
/*
 * Init SIP stack
 */
static pj_status_t init_sip()
{
    unsigned i;
    pj_status_t status;
 
    /* init PJLIB-UTIL: */
    status = pjlib_util_init();
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
    /* Must create a pool factory before we can allocate any memory. */
    pj_caching_pool_init(&app.cp, &pj_pool_factory_default_policy, 0);
 
    /* Create application pool for misc. */
    app.pool = pj_pool_create(&app.cp.factory, "app", 1000, 1000, NULL);
 
    /* Create the endpoint: */
    status = pjsip_endpt_create(&app.cp.factory, pj_gethostname()->ptr, 
                                &app.sip_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
 
    /* Add UDP transport. */
    {
        pj_sockaddr_in addr;
        pjsip_host_port addrname;
        pjsip_transport *tp;
 
        pj_bzero(&addr, sizeof(addr));
        addr.sin_family = pj_AF_INET();
        addr.sin_addr.s_addr = 0;
        addr.sin_port = pj_htons((pj_uint16_t)app.sip_port);
 
        if (app.local_addr.slen) {
 
            addrname.host = app.local_addr;
            addrname.port = app.sip_port;
 
            status = pj_sockaddr_in_init(&addr, &app.local_addr, 
                                         (pj_uint16_t)app.sip_port);
            if (status != PJ_SUCCESS) {
                app_perror(THIS_FILE, "Unable to resolve IP interface", status);
                return status;
            }
        }
 
        status = pjsip_udp_transport_start( app.sip_endpt, &addr, 
                                            (app.local_addr.slen ? &addrname:NULL),
                                            1, &tp);
        if (status != PJ_SUCCESS) {
            app_perror(THIS_FILE, "Unable to start UDP transport", status);
            return status;
        }
 
        PJ_LOG(3,(THIS_FILE, "SIP UDP listening on %.*s:%d",
                  (int)tp->local_name.host.slen, tp->local_name.host.ptr,
                  tp->local_name.port));
    }
 
    /* 
     * Init transaction layer.
     * This will create/initialize transaction hash tables etc.
     */
    status = pjsip_tsx_layer_init_module(app.sip_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
    /*  Initialize UA layer. */
    status = pjsip_ua_init_module( app.sip_endpt, NULL );
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
    /* Initialize 100rel support */
    status = pjsip_100rel_init_module(app.sip_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
    /*  Init invite session module. */
    {
        pjsip_inv_callback inv_cb;
 
        /* Init the callback for INVITE session: */
        pj_bzero(&inv_cb, sizeof(inv_cb));
        inv_cb.on_state_changed = &call_on_state_changed;
        inv_cb.on_new_session = &call_on_forked;
        inv_cb.on_media_update = &call_on_media_update;
 
        /* Initialize invite session module:  */
        status = pjsip_inv_usage_init(app.sip_endpt, &inv_cb);
        PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    }
 
    /* Register our module to receive incoming requests. */
    status = pjsip_endpt_register_module( app.sip_endpt, &mod_siprtp);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
    /* Init calls */
    for (i=0; i<app.max_calls; ++i)
        app.call[i].index = i;
 
    /* Done */
    return PJ_SUCCESS;
}
 
 
/*
 * Destroy SIP
 */
static void destroy_sip()
{
    unsigned i;
 
    app.thread_quit = 1;
    for (i=0; i<app.thread_count; ++i) {
        if (app.sip_thread[i]) {
            pj_thread_join(app.sip_thread[i]);
            pj_thread_destroy(app.sip_thread[i]);
            app.sip_thread[i] = NULL;
        }
    }
 
    if (app.sip_endpt) {
        pjsip_endpt_destroy(app.sip_endpt);
        app.sip_endpt = NULL;
    }
 
}
 
 
/*
 * Init media stack.
 */
static pj_status_t init_media()
{
    unsigned    i, count;
    pj_uint16_t rtp_port;
    pj_status_t status;
 
 
    /* Initialize media endpoint so that at least error subsystem is properly
     * initialized.
     */
#if PJ_HAS_THREADS
    status = pjmedia_endpt_create(&app.cp.factory, NULL, 1, &app.med_endpt);
#else
    status = pjmedia_endpt_create(&app.cp.factory, 
                                  pjsip_endpt_get_ioqueue(app.sip_endpt),
                                  0, &app.med_endpt);
#endif
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
 
    /* Must register codecs to be supported */
#if defined(PJMEDIA_HAS_G711_CODEC) && PJMEDIA_HAS_G711_CODEC!=0
    pjmedia_codec_g711_init(app.med_endpt);
#endif
 
    /* RTP port counter */
    rtp_port = (pj_uint16_t)(app.rtp_start_port & 0xFFFE);
 
    /* Init media transport for all calls. */
    for (i=0, count=0; i<app.max_calls; ++i, ++count) {
 
        unsigned j;
 
        /* Create transport for each media in the call */
        for (j=0; j<PJ_ARRAY_SIZE(app.call[0].media); ++j) {
            /* Repeat binding media socket to next port when fails to bind
             * to current port number.
             */
            int retry;
 
            app.call[i].media[j].call_index = i;
            app.call[i].media[j].media_index = j;
 
            status = -1;
            for (retry=0; retry<100; ++retry,rtp_port+=2)  {
                struct media_stream *m = &app.call[i].media[j];
                
                status = pjmedia_transport_udp_create2(app.med_endpt, 
                                                       "siprtp",
                                                       &app.local_addr,
                                                       rtp_port, 0, 
                                                       &m->transport);
                if (status == PJ_SUCCESS) {
                    rtp_port += 2;
                    break;
                }
            }
        }
 
        if (status != PJ_SUCCESS)
            goto on_error;
    }
 
    /* Done */
    return PJ_SUCCESS;
 
on_error:
    destroy_media();
    return status;
}
 
 
/*
 * Destroy media.
 */
static void destroy_media()
{
    unsigned i;
 
    for (i=0; i<app.max_calls; ++i) {
        unsigned j;
        for (j=0; j<PJ_ARRAY_SIZE(app.call[0].media); ++j) {
            struct media_stream *m = &app.call[i].media[j];
 
            if (m->transport) {
                pjmedia_transport_close(m->transport);
                m->transport = NULL;
            }
        }
    }
 
    if (app.med_endpt) {
        pjmedia_endpt_destroy(app.med_endpt);
        app.med_endpt = NULL;
    }
}
 
 
/*
 * Make outgoing call.
 */
static pj_status_t make_call(const pj_str_t *dst_uri)
{
    unsigned i;
    struct call *call;
    pjsip_dialog *dlg;
    pjmedia_sdp_session *sdp;
    pjsip_tx_data *tdata;
    pj_status_t status;
 
 
    /* Find unused call slot */
    for (i=0; i<app.max_calls; ++i) {
        if (app.call[i].inv == NULL)
            break;
    }
 
    if (i == app.max_calls)
        return PJ_ETOOMANY;
 
    call = &app.call[i];
 
    /* Create UAC dialog */
    status = pjsip_dlg_create_uac( pjsip_ua_instance(), 
                                   &app.local_uri,      /* local URI        */
                                   &app.local_contact,  /* local Contact    */
                                   dst_uri,             /* remote URI       */
                                   dst_uri,             /* remote target    */
                                   &dlg);               /* dialog           */
    if (status != PJ_SUCCESS) {
        ++app.uac_calls;
        return status;
    }
 
    /* Create SDP */
    create_sdp( dlg->pool, call, &sdp);
 
    /* Create the INVITE session. */
    status = pjsip_inv_create_uac( dlg, sdp, 0, &call->inv);
    if (status != PJ_SUCCESS) {
        pjsip_dlg_terminate(dlg);
        ++app.uac_calls;
        return status;
    }
 
 
    /* Attach call data to invite session */
    call->inv->mod_data[mod_siprtp.id] = call;
 
    /* Mark start of call */
    pj_gettimeofday(&call->start_time);
 
 
    /* Create initial INVITE request.
     * This INVITE request will contain a perfectly good request and 
     * an SDP body as well.
     */
    status = pjsip_inv_invite(call->inv, &tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
 
    /* Send initial INVITE request. 
     * From now on, the invite session's state will be reported to us
     * via the invite session callbacks.
     */
    status = pjsip_inv_send_msg(call->inv, tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
 
    return PJ_SUCCESS;
}
 
 
/*
 * Receive incoming call
 */
static void process_incoming_call(pjsip_rx_data *rdata)
{
    unsigned i, options;
    struct call *call;
    pjsip_dialog *dlg;
    pjmedia_sdp_session *sdp;
    pjsip_tx_data *tdata;
    pj_status_t status;
 
    /* Find free call slot */
    for (i=0; i<app.max_calls; ++i) {
        if (app.call[i].inv == NULL)
            break;
    }
 
    if (i == app.max_calls) {
        const pj_str_t reason = pj_str("Too many calls");
        pjsip_endpt_respond_stateless( app.sip_endpt, rdata, 
                                       500, &reason,
                                       NULL, NULL);
        return;
    }
 
    call = &app.call[i];
 
    /* Verify that we can handle the request. */
    options = 0;
    status = pjsip_inv_verify_request(rdata, &options, NULL, NULL,
                                   app.sip_endpt, &tdata);
    if (status != PJ_SUCCESS) {
        /*
         * No we can't handle the incoming INVITE request.
         */
        if (tdata) {
            pjsip_response_addr res_addr;
            
            pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
            status = pjsip_endpt_send_response(app.sip_endpt, &res_addr, tdata,
                                                                                   NULL, NULL);
            if (status != PJ_SUCCESS) pjsip_tx_data_dec_ref(tdata);
            
        } else {
            
            /* Respond with 500 (Internal Server Error) */
            pjsip_endpt_respond_stateless(app.sip_endpt, rdata, 500, NULL,
                NULL, NULL);
        }
        
        return;
    }
 
    /* Create UAS dialog */
    status = pjsip_dlg_create_uas_and_inc_lock( pjsip_ua_instance(), rdata,
                                                &app.local_contact, &dlg);
    if (status != PJ_SUCCESS) {
        const pj_str_t reason = pj_str("Unable to create dialog");
        pjsip_endpt_respond_stateless( app.sip_endpt, rdata, 
                                       500, &reason,
                                       NULL, NULL);
        return;
    }
 
    /* Create SDP */
    create_sdp( dlg->pool, call, &sdp);
 
    /* Create UAS invite session */
    status = pjsip_inv_create_uas( dlg, rdata, sdp, 0, &call->inv);
    if (status != PJ_SUCCESS) {
        pjsip_dlg_create_response(dlg, rdata, 500, NULL, &tdata);
        pjsip_dlg_send_response(dlg, pjsip_rdata_get_tsx(rdata), tdata);
        pjsip_dlg_dec_lock(dlg);
        return;
    }
    
    /* Invite session has been created, decrement & release dialog lock */
    pjsip_dlg_dec_lock(dlg);
 
    /* Attach call data to invite session */
    call->inv->mod_data[mod_siprtp.id] = call;
 
    /* Mark start of call */
    pj_gettimeofday(&call->start_time);
 
 
 
    /* Create 200 response .*/
    status = pjsip_inv_initial_answer(call->inv, rdata, 200, 
                                      NULL, NULL, &tdata);
    if (status != PJ_SUCCESS) {
        status = pjsip_inv_initial_answer(call->inv, rdata, 
                                          PJSIP_SC_NOT_ACCEPTABLE,
                                          NULL, NULL, &tdata);
        if (status == PJ_SUCCESS)
            pjsip_inv_send_msg(call->inv, tdata); 
        else
            pjsip_inv_terminate(call->inv, 500, PJ_FALSE);
        return;
    }
 
 
    /* Send the 200 response. */  
    status = pjsip_inv_send_msg(call->inv, tdata); 
    PJ_ASSERT_ON_FAIL(status == PJ_SUCCESS, return);
 
 
    /* Done */
}
 
 
/* Callback to be called when dialog has forked: */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e)
{
    PJ_UNUSED_ARG(inv);
    PJ_UNUSED_ARG(e);
 
    PJ_TODO( HANDLE_FORKING );
}
 
 
/* Callback to be called to handle incoming requests outside dialogs: */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata )
{
    /* Ignore strandled ACKs (must not send respone */
    if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD)
        return PJ_FALSE;
 
    /* Respond (statelessly) any non-INVITE requests with 500  */
    if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
        pj_str_t reason = pj_str("Unsupported Operation");
        pjsip_endpt_respond_stateless( app.sip_endpt, rdata, 
                                       500, &reason,
                                       NULL, NULL);
        return PJ_TRUE;
    }
 
    /* Handle incoming INVITE */
    process_incoming_call(rdata);
 
    /* Done */
    return PJ_TRUE;
}
 
 
/* Callback timer to disconnect call (limiting call duration) */
static void timer_disconnect_call( pj_timer_heap_t *timer_heap,
                                   struct pj_timer_entry *entry)
{
    struct call *call = (struct call *)(entry->user_data);
 
    PJ_UNUSED_ARG(timer_heap);
 
    entry->id = 0;
    hangup_call(call->index);
}
 
 
/* Callback to be called when invite session's state has changed: */
static void call_on_state_changed( pjsip_inv_session *inv, 
                                   pjsip_event *e)
{
    struct call *call = (struct call *)inv->mod_data[mod_siprtp.id];
 
    PJ_UNUSED_ARG(e);
 
    if (!call)
        return;
 
    if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
        
        pj_time_val null_time = {0, 0};
 
        if (call->d_timer.id != 0) {
            pjsip_endpt_cancel_timer(app.sip_endpt, &call->d_timer);
            call->d_timer.id = 0;
        }
 
        PJ_LOG(3,(THIS_FILE, "Call #%d disconnected. Reason=%d (%.*s)",
                  call->index,
                  inv->cause,
                  (int)inv->cause_text.slen,
                  inv->cause_text.ptr));
 
        if (app.call_report) {
            PJ_LOG(3,(THIS_FILE, "Call #%d statistics:", call->index));
            print_call(call->index);
        }
 
 
        call->inv = NULL;
        inv->mod_data[mod_siprtp.id] = NULL;
 
        destroy_call_media(call->index);
 
        call->start_time = null_time;
        call->response_time = null_time;
        call->connect_time = null_time;
 
        ++app.uac_calls;
 
    } else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
 
        pj_time_val t;
 
        pj_gettimeofday(&call->connect_time);
        if (call->response_time.sec == 0)
            call->response_time = call->connect_time;
 
        t = call->connect_time;
        PJ_TIME_VAL_SUB(t, call->start_time);
 
        PJ_LOG(3,(THIS_FILE, "Call #%d connected in %d ms", call->index,
                  PJ_TIME_VAL_MSEC(t)));
 
        if (app.duration != 0) {
            call->d_timer.id = 1;
            call->d_timer.user_data = call;
            call->d_timer.cb = &timer_disconnect_call;
 
            t.sec = app.duration;
            t.msec = 0;
 
            pjsip_endpt_schedule_timer(app.sip_endpt, &call->d_timer, &t);
        }
 
    } else if ( inv->state == PJSIP_INV_STATE_EARLY ||
                inv->state == PJSIP_INV_STATE_CONNECTING) {
 
        if (call->response_time.sec == 0)
            pj_gettimeofday(&call->response_time);
 
    }
}
 
 
/* Utility */
static void app_perror(const char *sender, const char *title, 
                       pj_status_t status)
{
    char errmsg[PJ_ERR_MSG_SIZE];
 
    pj_strerror(status, errmsg, sizeof(errmsg));
    PJ_LOG(3,(sender, "%s: %s [status=%d]", title, errmsg, status));
}
 
 
/* Worker thread for SIP */
static int sip_worker_thread(void *arg)
{
    PJ_UNUSED_ARG(arg);
 
    while (!app.thread_quit) {
        pj_time_val timeout = {0, 10};
        pjsip_endpt_handle_events(app.sip_endpt, &timeout);
    }
 
    return 0;
}
 
 
/* Init application options */
static pj_status_t init_options(int argc, char *argv[])
{
    static char ip_addr[PJ_INET_ADDRSTRLEN];
    static char local_uri[64];
 
    enum { OPT_START,
           OPT_APP_LOG_LEVEL, OPT_LOG_FILE, 
           OPT_A_PT, OPT_A_NAME, OPT_A_CLOCK, OPT_A_BITRATE, OPT_A_PTIME,
           OPT_REPORT_FILE };
 
    struct pj_getopt_option long_options[] = {
        { "count",          1, 0, 'c' },
        { "gap",            1, 0, 'g' },
        { "call-report",    0, 0, 'R' },
        { "duration",       1, 0, 'd' },
        { "auto-quit",      0, 0, 'q' },
        { "local-port",     1, 0, 'p' },
        { "rtp-port",       1, 0, 'r' },
        { "ip-addr",        1, 0, 'i' },
 
        { "log-level",      1, 0, 'l' },
        { "app-log-level",  1, 0, OPT_APP_LOG_LEVEL },
        { "log-file",       1, 0, OPT_LOG_FILE },
 
        { "report-file",    1, 0, OPT_REPORT_FILE },
 
        /* Don't support this anymore, see comments in USAGE above.
        { "a-pt",           1, 0, OPT_A_PT },
        { "a-name",         1, 0, OPT_A_NAME },
        { "a-clock",        1, 0, OPT_A_CLOCK },
        { "a-bitrate",      1, 0, OPT_A_BITRATE },
        { "a-ptime",        1, 0, OPT_A_PTIME },
        */
 
        { NULL, 0, 0, 0 },
    };
    int c;
    int option_index;
 
    /* Get local IP address for the default IP address */
    {
        const pj_str_t *hostname;
        pj_sockaddr_in tmp_addr;
 
        hostname = pj_gethostname();
        pj_sockaddr_in_init(&tmp_addr, hostname, 0);
        pj_inet_ntop(pj_AF_INET(), &tmp_addr.sin_addr, ip_addr,
                     sizeof(ip_addr));
    }
 
    /* Init defaults */
    app.max_calls = 1;
    app.thread_count = 1;
    app.sip_port = 5060;
    app.rtp_start_port = RTP_START_PORT;
    app.local_addr = pj_str(ip_addr);
    app.log_level = 5;
    app.app_log_level = 3;
    app.log_filename = NULL;
 
    /* Default codecs: */
    app.audio_codec = audio_codecs[0];
 
    /* Parse options */
    pj_optind = 0;
    while((c=pj_getopt_long(argc,argv, "c:d:p:r:i:l:g:qR", 
                            long_options, &option_index))!=-1) 
    {
        switch (c) {
        case 'c':
            app.max_calls = atoi(pj_optarg);
            if (app.max_calls > MAX_CALLS) {
                PJ_LOG(3,(THIS_FILE,"Invalid max calls value %s "
                                    "(must be <= %d)", pj_optarg, MAX_CALLS));
                return 1;
            }
            break;
        case 'g':
            app.call_gap = atoi(pj_optarg);
            break;
        case 'R':
            app.call_report = PJ_TRUE;
            break;
        case 'd':
            app.duration = atoi(pj_optarg);
            break;
        case 'q':
            app.auto_quit = 1;
            break;
 
        case 'p':
            app.sip_port = atoi(pj_optarg);
            break;
        case 'r':
            app.rtp_start_port = atoi(pj_optarg);
            break;
        case 'i':
            app.local_addr = pj_str(pj_optarg);
            break;
 
        case 'l':
            app.log_level = atoi(pj_optarg);
            break;
        case OPT_APP_LOG_LEVEL:
            app.app_log_level = atoi(pj_optarg);
            break;
        case OPT_LOG_FILE:
            app.log_filename = pj_optarg;
            break;
 
        case OPT_A_PT:
            app.audio_codec.pt = atoi(pj_optarg);
            break;
        case OPT_A_NAME:
            app.audio_codec.name = pj_optarg;
            break;
        case OPT_A_CLOCK:
            app.audio_codec.clock_rate = atoi(pj_optarg);
            break;
        case OPT_A_BITRATE:
            app.audio_codec.bit_rate = atoi(pj_optarg);
            break;
        case OPT_A_PTIME:
            app.audio_codec.ptime = atoi(pj_optarg);
            break;
        case OPT_REPORT_FILE:
            app.report_filename = pj_optarg;
            break;
 
        default:
            puts(USAGE);
            return 1;
        }
    }
 
    /* Check if URL is specified */
    if (pj_optind < argc)
        app.uri_to_call = pj_str(argv[pj_optind]);
 
    /* Build local URI and contact */
    pj_ansi_sprintf( local_uri, "sip:%s:%d", app.local_addr.ptr, app.sip_port);
    app.local_uri = pj_str(local_uri);
    app.local_contact = app.local_uri;
 
 
    return PJ_SUCCESS;
}
 
 
/*****************************************************************************
 * MEDIA STUFFS
 */
 
/*
 * Create SDP session for a call.
 */
static pj_status_t create_sdp( pj_pool_t *pool,
                               struct call *call,
                               pjmedia_sdp_session **p_sdp)
{
    pj_time_val tv;
    pjmedia_sdp_session *sdp;
    pjmedia_sdp_media *m;
    pjmedia_sdp_attr *attr;
    pjmedia_transport_info tpinfo;
    struct media_stream *audio = &call->media[0];
 
    PJ_ASSERT_RETURN(pool && p_sdp, PJ_EINVAL);
 
 
    /* Get transport info */
    pjmedia_transport_info_init(&tpinfo);
    pjmedia_transport_get_info(audio->transport, &tpinfo);
 
    /* Create and initialize basic SDP session */
    sdp = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_session);
 
    pj_gettimeofday(&tv);
    sdp->origin.user = pj_str("pjsip-siprtp");
    sdp->origin.version = sdp->origin.id = tv.sec + 2208988800UL;
    sdp->origin.net_type = pj_str("IN");
    sdp->origin.addr_type = pj_str("IP4");
    sdp->origin.addr = *pj_gethostname();
    sdp->name = pj_str("pjsip");
 
    /* Since we only support one media stream at present, put the
     * SDP connection line in the session level.
     */
    sdp->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
    sdp->conn->net_type = pj_str("IN");
    sdp->conn->addr_type = pj_str("IP4");
    sdp->conn->addr = app.local_addr;
 
 
    /* SDP time and attributes. */
    sdp->time.start = sdp->time.stop = 0;
    sdp->attr_count = 0;
 
    /* Create media stream 0: */
 
    sdp->media_count = 1;
    m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
    sdp->media[0] = m;
 
    /* Standard media info: */
    m->desc.media = pj_str("audio");
    m->desc.port = pj_ntohs(tpinfo.sock_info.rtp_addr_name.ipv4.sin_port);
    m->desc.port_count = 1;
    m->desc.transport = pj_str("RTP/AVP");
 
    /* Add format and rtpmap for each codec. */
    m->desc.fmt_count = 1;
    m->attr_count = 0;
 
    {
        pjmedia_sdp_rtpmap rtpmap;
        char ptstr[10];
 
        sprintf(ptstr, "%d", app.audio_codec.pt);
        pj_strdup2(pool, &m->desc.fmt[0], ptstr);
        rtpmap.pt = m->desc.fmt[0];
        rtpmap.clock_rate = app.audio_codec.clock_rate;
        rtpmap.enc_name = pj_str(app.audio_codec.name);
        rtpmap.param.slen = 0;
 
        pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
        m->attr[m->attr_count++] = attr;
    }
 
    /* Add sendrecv attribute. */
    attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
    attr->name = pj_str("sendrecv");
    m->attr[m->attr_count++] = attr;
 
#if 1
    /*
     * Add support telephony event
     */
    m->desc.fmt[m->desc.fmt_count++] = pj_str("121");
    /* Add rtpmap. */
    attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
    attr->name = pj_str("rtpmap");
    attr->value = pj_str("121 telephone-event/8000");
    m->attr[m->attr_count++] = attr;
    /* Add fmtp */
    attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
    attr->name = pj_str("fmtp");
    attr->value = pj_str("121 0-15");
    m->attr[m->attr_count++] = attr;
#endif
 
    /* Done */
    *p_sdp = sdp;
 
    return PJ_SUCCESS;
}
 
 
#if (defined(PJ_WIN32) && PJ_WIN32 != 0) || (defined(PJ_WIN64) && PJ_WIN64 != 0)
#include <windows.h>
static void boost_priority(void)
{
    SetPriorityClass( GetCurrentProcess(), REALTIME_PRIORITY_CLASS);
    SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
}
 
#elif defined(PJ_LINUX) && PJ_LINUX != 0
#include <pthread.h>
static void boost_priority(void)
{
#define POLICY  SCHED_FIFO
    struct sched_param tp;
    int max_prio;
    int policy;
    int rc;
 
    if (sched_get_priority_min(POLICY) < sched_get_priority_max(POLICY))
        max_prio = sched_get_priority_max(POLICY)-1;
    else
        max_prio = sched_get_priority_max(POLICY)+1;
 
    /*
     * Adjust process scheduling algorithm and priority
     */
    rc = sched_getparam(0, &tp);
    if (rc != 0) {
        app_perror( THIS_FILE, "sched_getparam error",
                    PJ_RETURN_OS_ERROR(rc));
        return;
    }
    tp.sched_priority = max_prio;
 
    rc = sched_setscheduler(0, POLICY, &tp);
    if (rc != 0) {
        app_perror( THIS_FILE, "sched_setscheduler error",
                    PJ_RETURN_OS_ERROR(rc));
    }
 
    PJ_LOG(4, (THIS_FILE, "New process policy=%d, priority=%d",
              policy, tp.sched_priority));
 
    /*
     * Adjust thread scheduling algorithm and priority
     */
    rc = pthread_getschedparam(pthread_self(), &policy, &tp);
    if (rc != 0) {
        app_perror( THIS_FILE, "pthread_getschedparam error",
                    PJ_RETURN_OS_ERROR(rc));
        return;
    }
 
    PJ_LOG(4, (THIS_FILE, "Old thread policy=%d, priority=%d",
              policy, tp.sched_priority));
 
    policy = POLICY;
    tp.sched_priority = max_prio;
 
    rc = pthread_setschedparam(pthread_self(), policy, &tp);
    if (rc != 0) {
        app_perror( THIS_FILE, "pthread_setschedparam error",
                    PJ_RETURN_OS_ERROR(rc));
        return;
    }
 
    PJ_LOG(4, (THIS_FILE, "New thread policy=%d, priority=%d",
              policy, tp.sched_priority));
}
 
#else
#  define boost_priority()
#endif
 
 
/*
 * This callback is called by media transport on receipt of RTP packet.
 */
static void on_rx_rtp(void *user_data, void *pkt, pj_ssize_t size)
{
    struct media_stream *strm;
    pj_status_t status;
    const pjmedia_rtp_hdr *hdr;
    const void *payload;
    unsigned payload_len;
 
    strm = (struct media_stream *)user_data;
 
    /* Discard packet if media is inactive */
    if (!strm->active)
        return;
 
    /* Check for errors */
    if (size < 0) {
        app_perror(THIS_FILE, "RTP recv() error", (pj_status_t)-size);
        return;
    }
 
    /* Decode RTP packet. */
    status = pjmedia_rtp_decode_rtp(&strm->in_sess, 
                                    pkt, (int)size, 
                                    &hdr, &payload, &payload_len);
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "RTP decode error", status);
        return;
    }
 
    //PJ_LOG(4,(THIS_FILE, "Rx seq=%d", pj_ntohs(hdr->seq)));
 
    /* Update the RTCP session. */
    pjmedia_rtcp_rx_rtp(&strm->rtcp, pj_ntohs(hdr->seq),
                        pj_ntohl(hdr->ts), payload_len);
 
    /* Update RTP session */
    pjmedia_rtp_session_update(&strm->in_sess, hdr, NULL);
 
}
 
/*
 * This callback is called by media transport on receipt of RTCP packet.
 */
static void on_rx_rtcp(void *user_data, void *pkt, pj_ssize_t size)
{
    struct media_stream *strm;
 
    strm = (struct media_stream *)user_data;
 
    /* Discard packet if media is inactive */
    if (!strm->active)
        return;
 
    /* Check for errors */
    if (size < 0) {
        app_perror(THIS_FILE, "Error receiving RTCP packet",(pj_status_t)-size);
        return;
    }
 
    /* Update RTCP session */
    pjmedia_rtcp_rx_rtcp(&strm->rtcp, pkt, size);
}
 
 
/* 
 * Media thread 
 *
 * This is the thread to send and receive both RTP and RTCP packets.
 */
static int media_thread(void *arg)
{
    enum { RTCP_INTERVAL = 5000, RTCP_RAND = 2000 };
    struct media_stream *strm = (struct media_stream *)arg;
    char packet[1500];
    unsigned msec_interval;
    pj_timestamp freq, next_rtp, next_rtcp;
 
 
    /* Boost thread priority if necessary */
    boost_priority();
 
    /* Let things settle */
    pj_thread_sleep(100);
 
    msec_interval = strm->samples_per_frame * 1000 / strm->clock_rate;
    pj_get_timestamp_freq(&freq);
 
    pj_get_timestamp(&next_rtp);
    next_rtp.u64 += (freq.u64 * msec_interval / 1000);
 
    next_rtcp = next_rtp;
    next_rtcp.u64 += (freq.u64 * (RTCP_INTERVAL+(pj_rand()%RTCP_RAND)) / 1000);
 
 
    while (!strm->thread_quit_flag) {
        pj_timestamp now, lesser;
        pj_time_val timeout;
        pj_bool_t send_rtp, send_rtcp;
 
        send_rtp = send_rtcp = PJ_FALSE;
 
        /* Determine how long to sleep */
        if (next_rtp.u64 < next_rtcp.u64) {
            lesser = next_rtp;
            send_rtp = PJ_TRUE;
        } else {
            lesser = next_rtcp;
            send_rtcp = PJ_TRUE;
        }
 
        pj_get_timestamp(&now);
        if (lesser.u64 <= now.u64) {
            timeout.sec = timeout.msec = 0;
            //printf("immediate "); fflush(stdout);
        } else {
            pj_uint64_t tick_delay;
            tick_delay = lesser.u64 - now.u64;
            timeout.sec = 0;
            timeout.msec = (pj_uint32_t)(tick_delay * 1000 / freq.u64);
            pj_time_val_normalize(&timeout);
 
            //printf("%d:%03d ", timeout.sec, timeout.msec); fflush(stdout);
        }
 
        /* Wait for next interval */
        //if (timeout.sec!=0 && timeout.msec!=0) {
            pj_thread_sleep(PJ_TIME_VAL_MSEC(timeout));
            if (strm->thread_quit_flag)
                break;
        //}
 
        pj_get_timestamp(&now);
 
        if (send_rtp || next_rtp.u64 <= now.u64) {
            /*
             * Time to send RTP packet.
             */
            pj_status_t status;
            const void *p_hdr;
            const pjmedia_rtp_hdr *hdr;
            pj_ssize_t size;
            int hdrlen;
 
            /* Format RTP header */
            status = pjmedia_rtp_encode_rtp( &strm->out_sess, strm->si.tx_pt,
                                             0, /* marker bit */
                                             strm->bytes_per_frame, 
                                             strm->samples_per_frame,
                                             &p_hdr, &hdrlen);
            if (status == PJ_SUCCESS) {
 
                //PJ_LOG(4,(THIS_FILE, "\t\tTx seq=%d", pj_ntohs(hdr->seq)));
                
                hdr = (const pjmedia_rtp_hdr*) p_hdr;
 
                /* Copy RTP header to packet */
                pj_memcpy(packet, hdr, hdrlen);
 
                /* Zero the payload */
                pj_bzero(packet+hdrlen, strm->bytes_per_frame);
 
                /* Send RTP packet */
                size = hdrlen + strm->bytes_per_frame;
                status = pjmedia_transport_send_rtp(strm->transport, 
                                                    packet, size);
                if (status != PJ_SUCCESS)
                    app_perror(THIS_FILE, "Error sending RTP packet", status);
 
            } else {
                pj_assert(!"RTP encode() error");
            }
 
            /* Update RTCP SR */
            pjmedia_rtcp_tx_rtp( &strm->rtcp, (pj_uint16_t)strm->bytes_per_frame);
 
            /* Schedule next send */
            next_rtp.u64 += (msec_interval * freq.u64 / 1000);
        }
 
 
        if (send_rtcp || next_rtcp.u64 <= now.u64) {
            /*
             * Time to send RTCP packet.
             */
            void *rtcp_pkt;
            int rtcp_len;
            pj_ssize_t size;
            pj_status_t status;
 
            /* Build RTCP packet */
            pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len);
 
    
            /* Send packet */
            size = rtcp_len;
            status = pjmedia_transport_send_rtcp(strm->transport,
                                                 rtcp_pkt, size);
            if (status != PJ_SUCCESS) {
                app_perror(THIS_FILE, "Error sending RTCP packet", status);
            }
            
            /* Schedule next send */
            next_rtcp.u64 += (freq.u64 * (RTCP_INTERVAL+(pj_rand()%RTCP_RAND)) /
                              1000);
        }
    }
 
    return 0;
}
 
 
/* Callback to be called when SDP negotiation is done in the call: */
static void call_on_media_update( pjsip_inv_session *inv,
                                  pj_status_t status)
{
    struct call *call;
    struct media_stream *audio;
    const pjmedia_sdp_session *local_sdp, *remote_sdp;
    struct codec *codec_desc = NULL;
    unsigned i;
 
    call = (struct call *)inv->mod_data[mod_siprtp.id];
    audio = &call->media[0];
 
    /* If this is a mid-call media update, then destroy existing media */
    if (audio->thread != NULL)
        destroy_call_media(call->index);
 
 
    /* Do nothing if media negotiation has failed */
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "SDP negotiation failed", status);
        return;
    }
 
    
    /* Capture stream definition from the SDP */
    pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
    pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
 
    status = pjmedia_stream_info_from_sdp(&audio->si, inv->pool, app.med_endpt,
                                          local_sdp, remote_sdp, 0);
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "Error creating stream info from SDP", status);
        return;
    }
 
    /* Get the remainder of codec information from codec descriptor */
    if (audio->si.fmt.pt == app.audio_codec.pt)
        codec_desc = &app.audio_codec;
    else {
        /* Find the codec description in codec array */
        for (i=0; i<PJ_ARRAY_SIZE(audio_codecs); ++i) {
            if (audio_codecs[i].pt == audio->si.fmt.pt) {
                codec_desc = &audio_codecs[i];
                break;
            }
        }
 
        if (codec_desc == NULL) {
            PJ_LOG(3, (THIS_FILE, "Error: Invalid codec payload type"));
            return;
        }
    }
 
    audio->clock_rate = audio->si.fmt.clock_rate;
    audio->samples_per_frame = audio->clock_rate * codec_desc->ptime / 1000;
    audio->bytes_per_frame = codec_desc->bit_rate * codec_desc->ptime / 1000 / 8;
 
 
    pjmedia_rtp_session_init(&audio->out_sess, audio->si.tx_pt, 
                             pj_rand());
    pjmedia_rtp_session_init(&audio->in_sess, audio->si.fmt.pt, 0);
    pjmedia_rtcp_init(&audio->rtcp, "rtcp", audio->clock_rate, 
                      audio->samples_per_frame, 0);
 
 
    /* Attach media to transport */
    status = pjmedia_transport_attach(audio->transport, audio, 
                                      &audio->si.rem_addr, 
                                      &audio->si.rem_rtcp, 
                                      sizeof(pj_sockaddr_in),
                                      &on_rx_rtp,
                                      &on_rx_rtcp);
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "Error on pjmedia_transport_attach()", status);
        return;
    }
 
    /* Start media transport */
    pjmedia_transport_media_start(audio->transport, 0, 0, 0, 0);
 
    /* Start media thread. */
    audio->thread_quit_flag = 0;
#if PJ_HAS_THREADS
    status = pj_thread_create( inv->pool, "media", &media_thread, audio,
                               0, 0, &audio->thread);
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "Error creating media thread", status);
        return;
    }
#endif
 
    /* Set the media as active */
    audio->active = PJ_TRUE;
}
 
 
 
/* Destroy call's media */
static void destroy_call_media(unsigned call_index)
{
    struct media_stream *audio = &app.call[call_index].media[0];
 
    if (audio) {
        audio->active = PJ_FALSE;
 
        if (audio->thread) {
            audio->thread_quit_flag = 1;
            pj_thread_join(audio->thread);
            pj_thread_destroy(audio->thread);
            audio->thread = NULL;
            audio->thread_quit_flag = 0;
        }
 
        pjmedia_transport_detach(audio->transport, audio);
    }
}
 
 
/*****************************************************************************
 * USER INTERFACE STUFFS
 */
 
static void call_get_duration(int call_index, pj_time_val *dur)
{
    struct call *call = &app.call[call_index];
    pjsip_inv_session *inv;
 
    dur->sec = dur->msec = 0;
 
    if (!call)
        return;
 
    inv = call->inv;
    if (!inv)
        return;
 
    if (inv->state >= PJSIP_INV_STATE_CONFIRMED && call->connect_time.sec) {
 
        pj_gettimeofday(dur);
        PJ_TIME_VAL_SUB((*dur), call->connect_time);
    }
}
 
 
static const char *good_number(char *buf, pj_int32_t val)
{
    if (val < 1000) {
        pj_ansi_sprintf(buf, "%d", val);
    } else if (val < 1000000) {
        pj_ansi_sprintf(buf, "%d.%02dK", 
                        val / 1000,
                        (val % 1000) / 100);
    } else {
        pj_ansi_sprintf(buf, "%d.%02dM", 
                        val / 1000000,
                        (val % 1000000) / 10000);
    }
 
    return buf;
}
 
 
 
static void print_avg_stat(void)
{
#define MIN_(var,val)      if ((int)val < (int)var) var = val
#define MAX_(var,val)      if ((int)val > (int)var) var = val
#define AVG_(var,val)      var = ( ((var * count) + val) / (count+1) )
#define BIGVAL              0x7FFFFFFFL
    struct stat_entry
    {
        int min, avg, max;
    };
 
    struct stat_entry call_dur, call_pdd;
    pjmedia_rtcp_stat min_stat, avg_stat, max_stat;
 
    char srx_min[16], srx_avg[16], srx_max[16];
    char brx_min[16], brx_avg[16], brx_max[16];
    char stx_min[16], stx_avg[16], stx_max[16];
    char btx_min[16], btx_avg[16], btx_max[16];
 
 
    unsigned i, count;
 
    pj_bzero(&call_dur, sizeof(call_dur)); 
    call_dur.min = BIGVAL;
 
    pj_bzero(&call_pdd, sizeof(call_pdd)); 
    call_pdd.min = BIGVAL;
 
    pj_bzero(&min_stat, sizeof(min_stat));
    min_stat.rx.pkt = min_stat.tx.pkt = BIGVAL;
    min_stat.rx.bytes = min_stat.tx.bytes = BIGVAL;
    min_stat.rx.loss = min_stat.tx.loss = BIGVAL;
    min_stat.rx.dup = min_stat.tx.dup = BIGVAL;
    min_stat.rx.reorder = min_stat.tx.reorder = BIGVAL;
    min_stat.rx.jitter.min = min_stat.tx.jitter.min = BIGVAL;
    min_stat.rtt.min = BIGVAL;
 
    pj_bzero(&avg_stat, sizeof(avg_stat));
    pj_bzero(&max_stat, sizeof(max_stat));
 
 
    for (i=0, count=0; i<app.max_calls; ++i) {
 
        struct call *call = &app.call[i];
        struct media_stream *audio = &call->media[0];
        pj_time_val dur;
        unsigned msec_dur;
 
        if (call->inv == NULL || 
            call->inv->state < PJSIP_INV_STATE_CONFIRMED ||
            call->connect_time.sec == 0) 
        {
            continue;
        }
 
        /* Duration */
        call_get_duration(i, &dur);
        msec_dur = PJ_TIME_VAL_MSEC(dur);
 
        MIN_(call_dur.min, msec_dur);
        MAX_(call_dur.max, msec_dur);
        AVG_(call_dur.avg, msec_dur);
 
        /* Connect delay */
        if (call->connect_time.sec) {
            pj_time_val t = call->connect_time;
            PJ_TIME_VAL_SUB(t, call->start_time);
            msec_dur = PJ_TIME_VAL_MSEC(t);
        } else {
            msec_dur = 10;
        }
 
        MIN_(call_pdd.min, msec_dur);
        MAX_(call_pdd.max, msec_dur);
        AVG_(call_pdd.avg, msec_dur);
 
        /* RX Statistisc: */
 
        /* Packets */
        MIN_(min_stat.rx.pkt, audio->rtcp.stat.rx.pkt);
        MAX_(max_stat.rx.pkt, audio->rtcp.stat.rx.pkt);
        AVG_(avg_stat.rx.pkt, audio->rtcp.stat.rx.pkt);
 
        /* Bytes */
        MIN_(min_stat.rx.bytes, audio->rtcp.stat.rx.bytes);
        MAX_(max_stat.rx.bytes, audio->rtcp.stat.rx.bytes);
        AVG_(avg_stat.rx.bytes, audio->rtcp.stat.rx.bytes);
 
 
        /* Packet loss */
        MIN_(min_stat.rx.loss, audio->rtcp.stat.rx.loss);
        MAX_(max_stat.rx.loss, audio->rtcp.stat.rx.loss);
        AVG_(avg_stat.rx.loss, audio->rtcp.stat.rx.loss);
 
        /* Packet dup */
        MIN_(min_stat.rx.dup, audio->rtcp.stat.rx.dup);
        MAX_(max_stat.rx.dup, audio->rtcp.stat.rx.dup);
        AVG_(avg_stat.rx.dup, audio->rtcp.stat.rx.dup);
 
        /* Packet reorder */
        MIN_(min_stat.rx.reorder, audio->rtcp.stat.rx.reorder);
        MAX_(max_stat.rx.reorder, audio->rtcp.stat.rx.reorder);
        AVG_(avg_stat.rx.reorder, audio->rtcp.stat.rx.reorder);
 
        /* Jitter  */
        MIN_(min_stat.rx.jitter.min, audio->rtcp.stat.rx.jitter.min);
        MAX_(max_stat.rx.jitter.max, audio->rtcp.stat.rx.jitter.max);
        AVG_(avg_stat.rx.jitter.mean, audio->rtcp.stat.rx.jitter.mean);
 
 
        /* TX Statistisc: */
 
        /* Packets */
        MIN_(min_stat.tx.pkt, audio->rtcp.stat.tx.pkt);
        MAX_(max_stat.tx.pkt, audio->rtcp.stat.tx.pkt);
        AVG_(avg_stat.tx.pkt, audio->rtcp.stat.tx.pkt);
 
        /* Bytes */
        MIN_(min_stat.tx.bytes, audio->rtcp.stat.tx.bytes);
        MAX_(max_stat.tx.bytes, audio->rtcp.stat.tx.bytes);
        AVG_(avg_stat.tx.bytes, audio->rtcp.stat.tx.bytes);
 
        /* Packet loss */
        MIN_(min_stat.tx.loss, audio->rtcp.stat.tx.loss);
        MAX_(max_stat.tx.loss, audio->rtcp.stat.tx.loss);
        AVG_(avg_stat.tx.loss, audio->rtcp.stat.tx.loss);
 
        /* Packet dup */
        MIN_(min_stat.tx.dup, audio->rtcp.stat.tx.dup);
        MAX_(max_stat.tx.dup, audio->rtcp.stat.tx.dup);
        AVG_(avg_stat.tx.dup, audio->rtcp.stat.tx.dup);
 
        /* Packet reorder */
        MIN_(min_stat.tx.reorder, audio->rtcp.stat.tx.reorder);
        MAX_(max_stat.tx.reorder, audio->rtcp.stat.tx.reorder);
        AVG_(avg_stat.tx.reorder, audio->rtcp.stat.tx.reorder);
 
        /* Jitter  */
        MIN_(min_stat.tx.jitter.min, audio->rtcp.stat.tx.jitter.min);
        MAX_(max_stat.tx.jitter.max, audio->rtcp.stat.tx.jitter.max);
        AVG_(avg_stat.tx.jitter.mean, audio->rtcp.stat.tx.jitter.mean);
 
 
        /* RTT */
        MIN_(min_stat.rtt.min, audio->rtcp.stat.rtt.min);
        MAX_(max_stat.rtt.max, audio->rtcp.stat.rtt.max);
        AVG_(avg_stat.rtt.mean, audio->rtcp.stat.rtt.mean);
 
        ++count;
    }
 
    if (count == 0) {
        puts("No active calls");
        return;
    }
 
    printf("Total %d call(s) active.\n"
           "                    Average Statistics\n"
           "                    min     avg     max \n"
           "                -----------------------\n"
           " call duration: %7d %7d %7d %s\n"
           " connect delay: %7d %7d %7d %s\n"
           " RX stat:\n"
           "       packets: %7s %7s %7s %s\n"
           "       payload: %7s %7s %7s %s\n"
           "          loss: %7d %7d %7d %s\n"
           "  percent loss: %7.3f %7.3f %7.3f %s\n"
           "           dup: %7d %7d %7d %s\n"
           "       reorder: %7d %7d %7d %s\n"
           "        jitter: %7.3f %7.3f %7.3f %s\n"
           " TX stat:\n"
           "       packets: %7s %7s %7s %s\n"
           "       payload: %7s %7s %7s %s\n"
           "          loss: %7d %7d %7d %s\n"
           "  percent loss: %7.3f %7.3f %7.3f %s\n"
           "           dup: %7d %7d %7d %s\n"
           "       reorder: %7d %7d %7d %s\n"
           "        jitter: %7.3f %7.3f %7.3f %s\n"
           " RTT          : %7.3f %7.3f %7.3f %s\n"
           ,
           count,
           call_dur.min/1000, call_dur.avg/1000, call_dur.max/1000, 
           "seconds",
 
           call_pdd.min, call_pdd.avg, call_pdd.max, 
           "ms",
 
           /* rx */
 
           good_number(srx_min, min_stat.rx.pkt),
           good_number(srx_avg, avg_stat.rx.pkt),
           good_number(srx_max, max_stat.rx.pkt),
           "packets",
 
           good_number(brx_min, min_stat.rx.bytes),
           good_number(brx_avg, avg_stat.rx.bytes),
           good_number(brx_max, max_stat.rx.bytes),
           "bytes",
 
           min_stat.rx.loss, avg_stat.rx.loss, max_stat.rx.loss,
           "packets",
           
           min_stat.rx.loss*100.0/(min_stat.rx.pkt+min_stat.rx.loss),
           avg_stat.rx.loss*100.0/(avg_stat.rx.pkt+avg_stat.rx.loss),
           max_stat.rx.loss*100.0/(max_stat.rx.pkt+max_stat.rx.loss),
           "%",
 
 
           min_stat.rx.dup, avg_stat.rx.dup, max_stat.rx.dup,
           "packets",
 
           min_stat.rx.reorder, avg_stat.rx.reorder, max_stat.rx.reorder,
           "packets",
 
           min_stat.rx.jitter.min/1000.0, 
           avg_stat.rx.jitter.mean/1000.0, 
           max_stat.rx.jitter.max/1000.0,
           "ms",
        
           /* tx */
 
           good_number(stx_min, min_stat.tx.pkt),
           good_number(stx_avg, avg_stat.tx.pkt),
           good_number(stx_max, max_stat.tx.pkt),
           "packets",
 
           good_number(btx_min, min_stat.tx.bytes),
           good_number(btx_avg, avg_stat.tx.bytes),
           good_number(btx_max, max_stat.tx.bytes),
           "bytes",
 
           min_stat.tx.loss, avg_stat.tx.loss, max_stat.tx.loss,
           "packets",
           
           min_stat.tx.loss*100.0/(min_stat.tx.pkt+min_stat.tx.loss),
           avg_stat.tx.loss*100.0/(avg_stat.tx.pkt+avg_stat.tx.loss),
           max_stat.tx.loss*100.0/(max_stat.tx.pkt+max_stat.tx.loss),
           "%",
 
           min_stat.tx.dup, avg_stat.tx.dup, max_stat.tx.dup,
           "packets",
 
           min_stat.tx.reorder, avg_stat.tx.reorder, max_stat.tx.reorder,
           "packets",
 
           min_stat.tx.jitter.min/1000.0, 
           avg_stat.tx.jitter.mean/1000.0, 
           max_stat.tx.jitter.max/1000.0,
           "ms",
 
           /* rtt */
           min_stat.rtt.min/1000.0, 
           avg_stat.rtt.mean/1000.0, 
           max_stat.rtt.max/1000.0,
           "ms"
           );
 
}
 
 
#include "siprtp_report.c"
 
 
static void list_calls()
{
    unsigned i;
    puts("List all calls:");
    for (i=0; i<app.max_calls; ++i) {
        if (!app.call[i].inv)
            continue;
        print_call(i);
    }
}
 
static void hangup_call(unsigned index)
{
    pjsip_tx_data *tdata;
    pj_status_t status;
 
    if (app.call[index].inv == NULL)
        return;
 
    status = pjsip_inv_end_session(app.call[index].inv, 603, NULL, &tdata);
    if (status==PJ_SUCCESS && tdata!=NULL)
        pjsip_inv_send_msg(app.call[index].inv, tdata);
}
 
static void hangup_all_calls()
{
    unsigned i;
    for (i=0; i<app.max_calls; ++i) {
        if (!app.call[i].inv)
            continue;
        hangup_call(i);
        pj_thread_sleep(app.call_gap);
    }
    
    /* Wait until all calls are terminated */
    for (i=0; i<app.max_calls; ++i) {
        while (app.call[i].inv)
            pj_thread_sleep(10);
    }
}
 
static pj_bool_t simple_input(const char *title, char *buf, pj_size_t len)
{
    char *p;
 
    printf("%s (empty to cancel): ", title); fflush(stdout);
    if (fgets(buf, (int)len, stdin) == NULL)
        return PJ_FALSE;
 
    /* Remove trailing newlines. */
    for (p=buf; ; ++p) {
        if (*p=='\r' || *p=='\n') *p='\0';
        else if (!*p) break;
    }
 
    if (!*buf)
        return PJ_FALSE;
    
    return PJ_TRUE;
}
 
 
static const char *MENU =
"\n"
"Enter menu character:\n"
"  s    Summary\n"
"  l    List all calls\n"
"  h    Hangup a call\n"
"  H    Hangup all calls\n"
"  q    Quit\n"
"\n";
 
 
/* Main screen menu */
static void console_main()
{
    char input1[10];
    unsigned i;
 
    printf("%s", MENU);
 
    for (;;) {
        printf(">>> "); fflush(stdout);
        if (fgets(input1, sizeof(input1), stdin) == NULL) {
            puts("EOF while reading stdin, will quit now..");
            break;
        }
 
        switch (input1[0]) {
 
        case 's':
            print_avg_stat();
            break;
 
        case 'l':
            list_calls();
            break;
 
        case 'h':
            if (!simple_input("Call number to hangup", input1, sizeof(input1)))
                break;
 
            i = atoi(input1);
            hangup_call(i);
            break;
 
        case 'H':
            hangup_all_calls();
            break;
 
        case 'q':
            goto on_exit;
 
        default:
            puts("Invalid command");
            printf("%s", MENU);
            break;
        }
 
        fflush(stdout);
    }
 
on_exit:
    hangup_all_calls();
}
 
 
/*****************************************************************************
 * Below is a simple module to log all incoming and outgoing SIP messages
 */
 
 
/* Notification on incoming messages */
static pj_bool_t logger_on_rx_msg(pjsip_rx_data *rdata)
{
    PJ_LOG(4,(THIS_FILE, "RX %d bytes %s from %s:%d:\n"
                         "%s\n"
                         "--end msg--",
                         rdata->msg_info.len,
                         pjsip_rx_data_get_info(rdata),
                         rdata->pkt_info.src_name,
                         rdata->pkt_info.src_port,
                         rdata->msg_info.msg_buf));
    
    /* Always return false, otherwise messages will not get processed! */
    return PJ_FALSE;
}
 
/* Notification on outgoing messages */
static pj_status_t logger_on_tx_msg(pjsip_tx_data *tdata)
{
    
    /* Important note:
     *  tp_info field is only valid after outgoing messages has passed
     *  transport layer. So don't try to access tp_info when the module
     *  has lower priority than transport layer.
     */
 
    PJ_LOG(4,(THIS_FILE, "TX %d bytes %s to %s:%d:\n"
                         "%s\n"
                         "--end msg--",
                         (tdata->buf.cur - tdata->buf.start),
                         pjsip_tx_data_get_info(tdata),
                         tdata->tp_info.dst_name,
                         tdata->tp_info.dst_port,
                         tdata->buf.start));
 
    /* Always return success, otherwise message will not get sent! */
    return PJ_SUCCESS;
}
 
/* The module instance. */
static pjsip_module msg_logger = 
{
    NULL, NULL,                         /* prev, next.          */
    { "mod-siprtp-log", 14 },           /* Name.                */
    -1,                                 /* Id                   */
    PJSIP_MOD_PRIORITY_TRANSPORT_LAYER-1,/* Priority            */
    NULL,                               /* load()               */
    NULL,                               /* start()              */
    NULL,                               /* stop()               */
    NULL,                               /* unload()             */
    &logger_on_rx_msg,                  /* on_rx_request()      */
    &logger_on_rx_msg,                  /* on_rx_response()     */
    &logger_on_tx_msg,                  /* on_tx_request.       */
    &logger_on_tx_msg,                  /* on_tx_response()     */
    NULL,                               /* on_tsx_state()       */
 
};
 
 
 
/*****************************************************************************
 * Console application custom logging:
 */
 
 
static FILE *log_file;
 
 
static void app_log_writer(int level, const char *buffer, int len)
{
    /* Write to both stdout and file. */
 
    if (level <= app.app_log_level)
        pj_log_write(level, buffer, len);
 
    if (log_file) {
        pj_size_t count = fwrite(buffer, len, 1, log_file);
        PJ_UNUSED_ARG(count);
        fflush(log_file);
    }
}
 
 
pj_status_t app_logging_init(void)
{
    /* Redirect log function to ours */
 
    pj_log_set_log_func( &app_log_writer );
 
    /* If output log file is desired, create the file: */
 
    if (app.log_filename) {
        log_file = fopen(app.log_filename, "wt");
        if (log_file == NULL) {
            PJ_LOG(1,(THIS_FILE, "Unable to open log file %s", 
                      app.log_filename));   
            return -1;
        }
    }
 
    return PJ_SUCCESS;
}
 
 
void app_logging_shutdown(void)
{
    /* Close logging file, if any: */
 
    if (log_file) {
        fclose(log_file);
        log_file = NULL;
    }
}
 
 
/*
 * main()
 */
int main(int argc, char *argv[])
{
    unsigned i;
    pj_status_t status;
 
    /* Must init PJLIB first */
    status = pj_init();
    if (status != PJ_SUCCESS)
        return 1;
 
    /* Get command line options */
    status = init_options(argc, argv);
    if (status != PJ_SUCCESS)
        return 1;
 
    /* Verify options: */
 
    /* Auto-quit can not be specified for UAS */
    if (app.auto_quit && app.uri_to_call.slen == 0) {
        printf("Error: --auto-quit option only valid for outgoing "
               "mode (UAC) only\n");
        return 1;
    }
 
    /* Init logging */
    status = app_logging_init();
    if (status != PJ_SUCCESS)
        return 1;
 
    /* Init SIP etc */
    status = init_sip();
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "Initialization has failed", status);
        destroy_sip();
        return 1;
    }
 
    /* Register module to log incoming/outgoing messages */
    pjsip_endpt_register_module(app.sip_endpt, &msg_logger);
 
    /* Init media */
    status = init_media();
    if (status != PJ_SUCCESS) {
        app_perror(THIS_FILE, "Media initialization failed", status);
        destroy_sip();
        return 1;
    }
 
    /* Start worker threads */
#if PJ_HAS_THREADS
    for (i=0; i<app.thread_count; ++i) {
        pj_thread_create( app.pool, "app", &sip_worker_thread, NULL,
                          0, 0, &app.sip_thread[i]);
    }
#endif
 
    /* If URL is specified, then make call immediately */
    if (app.uri_to_call.slen) {
        PJ_LOG(3,(THIS_FILE, "Making %d calls to %s..", app.max_calls,
                  app.uri_to_call.ptr));
 
        for (i=0; i<app.max_calls; ++i) {
            status = make_call(&app.uri_to_call);
            if (status != PJ_SUCCESS) {
                app_perror(THIS_FILE, "Error making call", status);
                break;
            }
            pj_thread_sleep(app.call_gap);
        }
 
        if (app.auto_quit) {
            /* Wait for calls to complete */
            while (app.uac_calls < app.max_calls)
                pj_thread_sleep(100);
            pj_thread_sleep(200);
        } else {
#if PJ_HAS_THREADS
            /* Start user interface loop */
            console_main();
#endif
        }
 
    } else {
 
        PJ_LOG(3,(THIS_FILE, "Ready for incoming calls (max=%d)", 
                  app.max_calls));
 
#if PJ_HAS_THREADS
        /* Start user interface loop */
        console_main();
#endif
    }
 
#if !PJ_HAS_THREADS
    PJ_LOG(3,(THIS_FILE, "Press Ctrl-C to quit"));
    for (;;) {
        pj_time_val t = {0, 10};
        pjsip_endpt_handle_events(app.sip_endpt, &t);
    }
#endif
    
    /* Shutting down... */
    destroy_sip();
    destroy_media();
 
    if (app.pool) {
        pj_pool_release(app.pool);
        app.pool = NULL;
        pj_caching_pool_destroy(&app.cp);
    }
 
    app_logging_shutdown();
 
    /* Shutdown PJLIB */
    pj_shutdown();
 
    return 0;
}