/* $Id$ */ /* * Copyright (C) 2003-2006 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 */ /* Include all headers. */ #include #include #include #include #include #include #include #include #define THIS_FILE "siprtp.c" #define MAX_CALLS 1024 #define RTP_START_PORT 44100 /* A bidirectional media stream */ struct media_stream { /* Static: */ pj_uint16_t port; /* RTP port (RTCP is +1) */ /* 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. */ /* Sockets: */ pj_sock_t rtp_sock; /* RTP socket. */ pj_sock_t rtcp_sock; /* RTCP socket. */ /* 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. */ pjmedia_rtcp_pkt rem_rtcp; /* received RTCP stat. */ /* Thread: */ pj_bool_t thread_quit_flag; /* worker thread quit flag */ pj_thread_t *thread; /* RTP/RTCP worker thread */ }; struct call { unsigned index; pjsip_inv_session *inv; unsigned media_count; struct media_stream media[2]; }; static struct app { unsigned max_calls; unsigned thread_count; int sip_port; int rtp_start_port; char *local_addr; pj_str_t local_uri; pj_str_t local_contact; 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 *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 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); /* Destroy the call's media */ static void destroy_call_media(unsigned call_index); /* Display error */ static void app_perror(const char *sender, const char *title, pj_status_t status); /* 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() */ }; /* * 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 global endpoint: */ { const pj_str_t *hostname; const char *endpt_name; /* Endpoint MUST be assigned a globally unique name. * The name will be used as the hostname in Warning header. */ /* For this implementation, we'll use hostname for simplicity */ hostname = pj_gethostname(); endpt_name = hostname->ptr; /* Create the endpoint: */ status = pjsip_endpt_create(&app.cp.factory, endpt_name, &app.sip_endpt); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); } /* Add UDP transport. */ { pj_sockaddr_in addr; addr.sin_family = PJ_AF_INET; addr.sin_addr.s_addr = 0; addr.sin_port = pj_htons((pj_uint16_t)app.sip_port); status = pjsip_udp_transport_start( app.sip_endpt, &addr, NULL, 1, NULL); if (status != PJ_SUCCESS) return status; } /* * 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); /* Init invite session module. */ { pjsip_inv_callback inv_cb; /* Init the callback for INVITE session: */ pj_memset(&inv_cb, 0, 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); /* Start worker threads */ for (i=0; iport = rtp_port; /* Create and bind RTP socket */ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &m->rtp_sock); if (status != PJ_SUCCESS) goto on_error; addr.sin_port = pj_htons(rtp_port); status = pj_sock_bind(m->rtp_sock, &addr, sizeof(addr)); if (status != PJ_SUCCESS) { pj_sock_close(m->rtp_sock), m->rtp_sock=0; continue; } /* Create and bind RTCP socket */ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &m->rtcp_sock); if (status != PJ_SUCCESS) goto on_error; addr.sin_port = pj_htons((pj_uint16_t)(rtp_port+1)); status = pj_sock_bind(m->rtcp_sock, &addr, sizeof(addr)); if (status != PJ_SUCCESS) { pj_sock_close(m->rtp_sock), m->rtp_sock=0; pj_sock_close(m->rtcp_sock), m->rtcp_sock=0; continue; } } while (status != PJ_SUCCESS && retry < 100); if (status != PJ_SUCCESS) goto on_error; } /* Done */ return PJ_SUCCESS; on_error: for (i=0; irtp_sock), m->rtp_sock=0; pj_sock_close(m->rtcp_sock), m->rtcp_sock=0; } return status; } /* * Destroy media. */ static void destroy_media() { unsigned i; for (i=0; irtp_sock) pj_sock_close(m->rtp_sock), m->rtp_sock = 0; if (m->rtcp_sock) pj_sock_close(m->rtcp_sock), m->rtcp_sock = 0; } 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); return status; } /* Attach call data to invite session */ call->inv->mod_data[mod_siprtp.id] = call; /* 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; 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, call, &sdp); /* Create UAS invite session */ status = pjsip_inv_create_uas( dlg, rdata, sdp, 0, &call->inv); if (status != PJ_SUCCESS) { pjsip_dlg_terminate(dlg); return; } /* Attach call data to invite session */ call->inv->mod_data[mod_siprtp.id] = call; /* Create 200 response .*/ status = pjsip_inv_initial_answer(call->inv, rdata, 200, NULL, NULL, &tdata); PJ_ASSERT_ON_FAIL(status == PJ_SUCCESS, 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 ) { /* 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 to be called when invite session's state has changed: */ static void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e) { PJ_UNUSED_ARG(e); if (inv->state == PJSIP_INV_STATE_DISCONNECTED) { struct call *call = inv->mod_data[mod_siprtp.id]; if (!call) return; call->inv = NULL; inv->mod_data[mod_siprtp.id] = NULL; destroy_call_media(call->index); } } /* 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 */ static int 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; } /* Usage */ static const char *USAGE = "Usage: \n" " siprtp [options] => to start in server mode \n" " siprtp [options] URL => to start in client mode \n" "\n" "where options are: \n" " --count=N, -c Set number of calls to create (default:1) \n" " --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" ; /* Init application options */ static pj_status_t init_options(int argc, char *argv[]) { static char ip_addr[32]; static char local_uri[64]; struct pj_getopt_option long_options[] = { { "count", 1, 0, 'c' }, { "port", 1, 0, 'p' }, { "rtp-port", 1, 0, 'r' }, { "ip-addr", 1, 0, 'i' }, { 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 default */ app.max_calls = 1; app.thread_count = 1; app.sip_port = 5060; app.rtp_start_port = 4000; app.local_addr = ip_addr; /* Parse options */ pj_optind = 0; while((c=pj_getopt_long(argc,argv, "c:p:r:i:", long_options, &option_index))!=-1) { switch (c) { case 'c': app.max_calls = atoi(pj_optarg); if (app.max_calls < 0 || app.max_calls > MAX_CALLS) { PJ_LOG(3,(THIS_FILE, "Invalid max calls value %s", pj_optarg)); return 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_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, 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; struct media_stream *audio = &call->media[0]; PJ_ASSERT_RETURN(pool && p_sdp, PJ_EINVAL); /* 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 = pj_str(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 = audio->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; PJ_TODO(PARAMETERIZE_CODEC); m->desc.fmt[0] = pj_str("0"); rtpmap.pt = pj_str("0"); rtpmap.clock_rate = 8000; rtpmap.enc_name = pj_str("pcmu"); 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("101"); /* Add rtpmap. */ attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr)); attr->name = pj_str("rtpmap"); attr->value = pj_str(":101 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(":101 0-15"); m->attr[m->attr_count++] = attr; #endif /* Done */ *p_sdp = sdp; return PJ_SUCCESS; } /* Media thread */ static int media_thread(void *arg) { struct media_stream *strm = arg; char packet[1500]; pj_time_val next_rtp, next_rtcp; pj_gettimeofday(&next_rtp); next_rtp.msec += strm->samples_per_frame * 1000 / strm->clock_rate; pj_time_val_normalize(&next_rtp); next_rtcp = next_rtp; next_rtcp.sec += 5; while (!strm->thread_quit_flag) { pj_fd_set_t set; pj_time_val now, lesser, timeout; int rc; /* Determine how long to sleep */ if (PJ_TIME_VAL_LT(next_rtp, next_rtcp)) lesser = next_rtp; else lesser = next_rtcp; pj_gettimeofday(&now); if (PJ_TIME_VAL_LTE(lesser, now)) timeout.sec = timeout.msec = 0; else { timeout = lesser; PJ_TIME_VAL_SUB(timeout, now); } PJ_FD_ZERO(&set); PJ_FD_SET(strm->rtp_sock, &set); PJ_FD_SET(strm->rtcp_sock, &set); rc = pj_sock_select(FD_SETSIZE, &set, NULL, NULL, &timeout); if (PJ_FD_ISSET(strm->rtp_sock, &set)) { /* * Process incoming RTP packet. */ pj_status_t status; pj_ssize_t size; const pjmedia_rtp_hdr *hdr; const void *payload; unsigned payload_len; size = sizeof(packet); status = pj_sock_recv(strm->rtp_sock, packet, &size, 0); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "RTP recv() error", status); continue; } /* Decode RTP packet. */ status = pjmedia_rtp_decode_rtp(&strm->in_sess, packet, size, &hdr, &payload, &payload_len); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "RTP decode error", status); continue; } /* Update RTP session */ status = pjmedia_rtp_session_update(&strm->in_sess, hdr); if (status != PJ_SUCCESS && status != PJMEDIA_RTP_ESESSPROBATION && status != PJMEDIA_RTP_ESESSRESTART) { app_perror(THIS_FILE, "RTP update error", status); PJ_LOG(3,(THIS_FILE,"RTP packet detail: pt=%d, seq=%d", hdr->pt, pj_ntohs(hdr->seq))); continue; } /* Update the RTCP session. */ pjmedia_rtcp_rx_rtp(&strm->rtcp, pj_ntohs(hdr->seq), pj_ntohl(hdr->ts)); } else if (PJ_FD_ISSET(strm->rtcp_sock, &set)) { /* * Process incoming RTCP */ pj_status_t status; pj_ssize_t size; size = sizeof(packet); status = pj_sock_recv( strm->rtcp_sock, packet, &size, 0); if (status != PJ_SUCCESS) app_perror(THIS_FILE, "Error receiving RTCP packet", status); else { if (size > sizeof(strm->rem_rtcp)) PJ_LOG(3,(THIS_FILE, "Error: RTCP packet too large")); else pj_memcpy(&strm->rem_rtcp, packet, size); } } pj_gettimeofday(&now); if (PJ_TIME_VAL_LTE(next_rtp, now)) { /* * Time to send RTP packet. */ pj_status_t status; 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, &hdr, &hdrlen); if (status == PJ_SUCCESS) { /* Copy RTP header to packet */ pj_memcpy(packet, hdr, hdrlen); /* Zero the payload */ pj_memset(packet+hdrlen, 0, strm->bytes_per_frame); /* Send RTP packet */ size = hdrlen + strm->bytes_per_frame; status = pj_sock_sendto( strm->rtp_sock, packet, &size, 0, &strm->si.rem_addr, sizeof(strm->si.rem_addr)); if (status != PJ_SUCCESS) app_perror(THIS_FILE, "Error sending RTP packet", status); } /* Update RTCP SR */ pjmedia_rtcp_tx_rtp( &strm->rtcp, (pj_uint16_t)strm->bytes_per_frame); /* Schedule next send */ next_rtp.msec += strm->samples_per_frame * 1000 / strm->clock_rate; pj_time_val_normalize(&next_rtp); } if (PJ_TIME_VAL_LTE(next_rtcp, now)) { /* * Time to send RTCP packet. */ pjmedia_rtcp_pkt *rtcp_pkt; int rtcp_len; pj_sockaddr_in rem_addr; pj_ssize_t size; int port; pj_status_t status; /* Build RTCP packet */ pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len); /* Calculate address based on RTP address */ rem_addr = strm->si.rem_addr; port = pj_ntohs(strm->si.rem_addr.sin_port) + 1; rem_addr.sin_port = pj_htons((pj_uint16_t)port); /* Send packet */ size = rtcp_len; status = pj_sock_sendto(strm->rtcp_sock, rtcp_pkt, &size, 0, &rem_addr, sizeof(rem_addr)); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Error sending RTCP packet", status); } next_rtcp.sec += 5; } } 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; pjmedia_sdp_session *local_sdp, *remote_sdp; 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; } audio->clock_rate = audio->si.fmt.sample_rate; audio->samples_per_frame = audio->clock_rate * 20 / 1000; audio->bytes_per_frame = 160; PJ_TODO(TAKE_CODEC_INFO_FROM_ARGUMENT); pjmedia_rtp_session_init(&audio->out_sess, audio->si.tx_pt, (pj_uint32_t)audio); pjmedia_rtp_session_init(&audio->in_sess, audio->si.fmt.pt, 0); pjmedia_rtcp_init(&audio->rtcp, 0); /* Start media thread. */ audio->thread_quit_flag = 0; 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); } } /* Destroy call's media */ static void destroy_call_media(unsigned call_index) { struct media_stream *audio = &app.call[call_index].media[0]; 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; } } ///////////////////////////////////////////////////////////////////////////// /* * USER INTERFACE STUFFS */ 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.%dK", val / 1000, (val % 1000) / 100); } else { pj_ansi_sprintf(buf, "%d.%02dM", val / 1000000, (val % 1000000) / 10000); } return buf; } static void print_call(int call_index) { int len; pjsip_inv_session *inv = app.call[call_index].inv; pjsip_dialog *dlg = inv->dlg; struct media_stream *audio = &app.call[call_index].media[0]; char userinfo[128]; char packets[16]; /* Dump invite sesion info. */ len = pjsip_hdr_print_on(dlg->remote.info, userinfo, sizeof(userinfo)); if (len < 1) pj_ansi_strcpy(userinfo, "<--uri too long-->"); else userinfo[len] = '\0'; printf("Call #%d: %s\n", call_index, pjsip_inv_state_name(inv->state)); printf(" %s\n", userinfo); if (app.call[call_index].media[0].thread == NULL) { return; } printf(" Stream #0: audio %.*s@%dHz, %d bytes/sec\n", (int)audio->si.fmt.encoding_name.slen, audio->si.fmt.encoding_name.ptr, audio->clock_rate, audio->bytes_per_frame * audio->clock_rate / audio->samples_per_frame); printf(" RX pkt=%s, fraction lost=%5.2f%%, jitter=%dms\n", good_number(packets, audio->rtcp.received), audio->rtcp.rtcp_pkt.rr.fract_lost/255.0, pj_ntohl(audio->rtcp.rtcp_pkt.rr.jitter) * 1000 / audio->clock_rate); printf(" TX pkt=%s, fraction lost=%5.2f%%, jitter=%dms\n", good_number(packets, pj_ntohl(audio->rtcp.rtcp_pkt.sr.sender_pcount)), audio->rem_rtcp.rr.fract_lost/255.0, pj_ntohl(audio->rem_rtcp.rr.jitter) * 1000 / audio->clock_rate); } static void list_calls() { unsigned i; puts("List all calls:"); for (i=0; i>> "); fflush(stdout); fgets(input1, sizeof(input1), stdin); switch (input1[0]) { 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: printf("%s", MENU); break; } fflush(stdout); } on_exit: ; } /* Notification on incoming messages */ static pj_bool_t console_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 console_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() */ &console_on_rx_msg, /* on_rx_request() */ &console_on_rx_msg, /* on_rx_response() */ &console_on_tx_msg, /* on_tx_request. */ &console_on_tx_msg, /* on_tx_response() */ NULL, /* on_tsx_state() */ }; /* * main() */ int main(int argc, char *argv[]) { pj_status_t status; status = pj_init(); if (status != PJ_SUCCESS) return 1; status = init_options(argc, argv); if (status != PJ_SUCCESS) return 1; status = init_sip(); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Initialization has failed", status); destroy_sip(); return 1; } pjsip_endpt_register_module(app.sip_endpt, &msg_logger); status = init_media(); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Media initialization failed", status); destroy_sip(); return 1; } if (app.uri_to_call.slen) { unsigned i; for (i=0; i