/* $Id$ */ /* * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 Benny Prijono * * 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 #include #include #include #include #include #include #include /* 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; /**< Disconnect 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; itransport); 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; itransport) { 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; ipool, 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; ipool, rdata, &res_addr); pjsip_endpt_send_response(app.sip_endpt, &res_addr, tdata, NULL, NULL); } 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( 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); return; } /* 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 = 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 = 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[32]; 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; char *addr; hostname = pj_gethostname(); pj_sockaddr_in_init(&tmp_addr, hostname, 0); addr = pj_inet_ntoa(tmp_addr.sin_addr); pj_ansi_strcpy(ip_addr, 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 (pool, sizeof(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 (pool, sizeof(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 (pool, sizeof(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; pjmedia_sdp_attr *attr; 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(pool, sizeof(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(pool, sizeof(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(pool, sizeof(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 static void boost_priority(void) { SetPriorityClass( GetCurrentProcess(), REALTIME_PRIORITY_CLASS); SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); } #elif defined(PJ_LINUX) && PJ_LINUX != 0 #include 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 = 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 = 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 = 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; pj_pool_t *pool; struct media_stream *audio; const pjmedia_sdp_session *local_sdp, *remote_sdp; struct codec *codec_desc = NULL; unsigned i; call = inv->mod_data[mod_siprtp.id]; pool = inv->dlg->pool; 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; isi.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 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; imedia[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>> "); 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