/* $Id$ */ /* * 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 */ /** * simpleua.c * * This is a very simple SIP user agent complete with media. The user * agent should do a proper SDP negotiation and start RTP media once * SDP negotiation has completed. * * This program does not register to SIP server. * * Capabilities to be demonstrated here: * - Basic call * - Should support IPv6 (not tested) * - UDP transport at port 5060 (hard coded) * - RTP socket at port 4000 (hard coded) * - proper SDP negotiation * - PCMA/PCMU codec only. * - Audio/media to sound device. * * * Usage: * - To make outgoing call, start simpleua with the URL of remote * destination to contact. * E.g.: * simpleua sip:user@remote * * - Incoming calls will automatically be answered with 180, then 200. * * This program does not disconnect call. * * This program will quit once it has completed a single call. */ /* Include all headers. */ #include #include #include #include #include #include #include /* For logging purpose. */ #define THIS_FILE "simpleua.c" #include "util.h" /* Settings */ #define AF pj_AF_INET() /* Change to pj_AF_INET6() for IPv6. * PJ_HAS_IPV6 must be enabled and * your system must support IPv6. */ #define SIP_PORT 5060 /* Listening SIP port */ #define RTP_PORT 4000 /* RTP port */ /* * Static variables. */ static pj_bool_t g_complete; /* Quit flag. */ static pjsip_endpoint *g_endpt; /* SIP endpoint. */ static pj_caching_pool cp; /* Global pool factory. */ static pjmedia_endpt *g_med_endpt; /* Media endpoint. */ static pjmedia_transport_info g_med_tpinfo; /* Socket info for media */ static pjmedia_transport *g_med_transport;/* Media stream transport */ /* Call variables: */ static pjsip_inv_session *g_inv; /* Current invite session. */ static pjmedia_session *g_med_session; /* Call's media session. */ static pjmedia_snd_port *g_snd_player; /* Call's sound player */ static pjmedia_snd_port *g_snd_rec; /* Call's sound recorder. */ /* * 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 ); /* 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_simpleua = { NULL, NULL, /* prev, next. */ { "mod-simpleua", 12 }, /* 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() */ }; /* * main() * * If called with argument, treat argument as SIP URL to be called. * Otherwise wait for incoming calls. */ int main(int argc, char *argv[]) { pj_status_t status; /* Must init PJLIB first: */ status = pj_init(); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* Then init PJLIB-UTIL: */ status = pjlib_util_init(); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* Must create a pool factory before we can allocate any memory. */ pj_caching_pool_init(&cp, &pj_pool_factory_default_policy, 0); /* 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(&cp.factory, endpt_name, &g_endpt); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); } /* * Add UDP transport, with hard-coded port * Alternatively, application can use pjsip_udp_transport_attach() to * start UDP transport, if it already has an UDP socket (e.g. after it * resolves the address with STUN). */ { pj_sockaddr addr; pj_sockaddr_init(AF, &addr, NULL, (pj_uint16_t)SIP_PORT); if (AF == pj_AF_INET()) { status = pjsip_udp_transport_start( g_endpt, &addr.ipv4, NULL, 1, NULL); } else if (AF == pj_AF_INET6()) { status = pjsip_udp_transport_start6(g_endpt, &addr.ipv6, NULL, 1, NULL); } else { status = PJ_EAFNOTSUP; } if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Unable to start UDP transport", status); return 1; } } /* * Init transaction layer. * This will create/initialize transaction hash tables etc. */ status = pjsip_tsx_layer_init_module(g_endpt); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* * Initialize UA layer module. * This will create/initialize dialog hash tables etc. */ status = pjsip_ua_init_module( g_endpt, NULL ); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* * Init invite session module. * The invite session module initialization takes additional argument, * i.e. a structure containing callbacks to be called on specific * occurence of events. * * The on_state_changed and on_new_session callbacks are mandatory. * Application must supply the callback function. * * We use on_media_update() callback in this application to start * media transmission. */ { 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(g_endpt, &inv_cb); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); } /* Initialize 100rel support */ status = pjsip_100rel_init_module(g_endpt); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); /* * Register our module to receive incoming requests. */ status = pjsip_endpt_register_module( g_endpt, &mod_simpleua); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* * Initialize media endpoint. * This will implicitly initialize PJMEDIA too. */ #if PJ_HAS_THREADS status = pjmedia_endpt_create(&cp.factory, NULL, 1, &g_med_endpt); #else status = pjmedia_endpt_create(&cp.factory, pjsip_endpt_get_ioqueue(g_endpt), 0, &g_med_endpt); #endif PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* * Add PCMA/PCMU codec to the media endpoint. */ #if defined(PJMEDIA_HAS_G711_CODEC) && PJMEDIA_HAS_G711_CODEC!=0 status = pjmedia_codec_g711_init(g_med_endpt); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); #endif /* * Create media transport used to send/receive RTP/RTCP socket. * One media transport is needed for each call. Application may * opt to re-use the same media transport for subsequent calls. */ status = pjmedia_transport_udp_create3(g_med_endpt, AF, NULL, NULL, RTP_PORT, 0, &g_med_transport); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Unable to create media transport", status); return 1; } /* * Get socket info (address, port) of the media transport. We will * need this info to create SDP (i.e. the address and port info in * the SDP). */ pjmedia_transport_info_init(&g_med_tpinfo); pjmedia_transport_get_info(g_med_transport, &g_med_tpinfo); /* * If URL is specified, then make call immediately. */ if (argc > 1) { pj_sockaddr hostaddr; char hostip[PJ_INET6_ADDRSTRLEN+2]; char temp[80]; pj_str_t dst_uri = pj_str(argv[1]); pj_str_t local_uri; pjsip_dialog *dlg; pjmedia_sdp_session *local_sdp; pjsip_tx_data *tdata; if (pj_gethostip(AF, &hostaddr) != PJ_SUCCESS) { app_perror(THIS_FILE, "Unable to retrieve local host IP", status); return 1; } pj_sockaddr_print(&hostaddr, hostip, sizeof(hostip), 2); pj_ansi_sprintf(temp, "", hostip, SIP_PORT); local_uri = pj_str(temp); /* Create UAC dialog */ status = pjsip_dlg_create_uac( pjsip_ua_instance(), &local_uri, /* local URI */ &local_uri, /* local Contact */ &dst_uri, /* remote URI */ &dst_uri, /* remote target */ &dlg); /* dialog */ if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Unable to create UAC dialog", status); return 1; } /* If we expect the outgoing INVITE to be challenged, then we should * put the credentials in the dialog here, with something like this: * { pjsip_cred_info cred[1]; cred[0].realm = pj_str("sip.server.realm"); cred[0].scheme = pj_str("digest"); cred[0].username = pj_str("theuser"); cred[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD; cred[0].data = pj_str("thepassword"); pjsip_auth_clt_set_credentials( &dlg->auth_sess, 1, cred); } * */ /* Get the SDP body to be put in the outgoing INVITE, by asking * media endpoint to create one for us. The SDP will contain all * codecs that have been registered to it (in this case, only * PCMA and PCMU), plus telephony event. */ status = pjmedia_endpt_create_sdp( g_med_endpt, /* the media endpt */ dlg->pool, /* pool. */ 1, /* # of streams */ &g_med_tpinfo.sock_info, /* RTP sock info */ &local_sdp); /* the SDP result */ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* Create the INVITE session, and pass the SDP returned earlier * as the session's initial capability. */ status = pjsip_inv_create_uac( dlg, local_sdp, 0, &g_inv); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* If we want the initial INVITE to travel to specific SIP proxies, * then we should put the initial dialog's route set here. The final * route set will be updated once a dialog has been established. * To set the dialog's initial route set, we do it with something * like this: * { pjsip_route_hdr route_set; pjsip_route_hdr *route; const pj_str_t hname = { "Route", 5 }; char *uri = "sip:proxy.server;lr"; pj_list_init(&route_set); route = pjsip_parse_hdr( dlg->pool, &hname, uri, strlen(uri), NULL); PJ_ASSERT_RETURN(route != NULL, 1); pj_list_push_back(&route_set, route); pjsip_dlg_set_route_set(dlg, &route_set); } * * Note that Route URI SHOULD have an ";lr" parameter! */ /* Create initial INVITE request. * This INVITE request will contain a perfectly good request and * an SDP body as well. */ status = pjsip_inv_invite(g_inv, &tdata); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* 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(g_inv, tdata); PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); } else { /* No URL to make call to */ PJ_LOG(3,(THIS_FILE, "Ready to accept incoming calls...")); } /* Loop until one call is completed */ for (;!g_complete;) { pj_time_val timeout = {0, 10}; pjsip_endpt_handle_events(g_endpt, &timeout); } /* On exit, dump current memory usage: */ dump_pool_usage(THIS_FILE, &cp); return 0; } /* * Callback when INVITE session state has changed. * This callback is registered when the invite session module is initialized. * We mostly want to know when the invite session has been disconnected, * so that we can quit the application. */ static void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e) { PJ_UNUSED_ARG(e); if (inv->state == PJSIP_INV_STATE_DISCONNECTED) { PJ_LOG(3,(THIS_FILE, "Call DISCONNECTED [reason=%d (%s)]", inv->cause, pjsip_get_status_text(inv->cause)->ptr)); PJ_LOG(3,(THIS_FILE, "One call completed, application quitting...")); g_complete = 1; } else { PJ_LOG(3,(THIS_FILE, "Call state changed to %s", pjsip_inv_state_name(inv->state))); } } /* This callback is called when dialog has forked. */ static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e) { /* To be done... */ PJ_UNUSED_ARG(inv); PJ_UNUSED_ARG(e); } /* * Callback when incoming requests outside any transactions and any * dialogs are received. We're only interested to hande incoming INVITE * request, and we'll reject any other requests with 500 response. */ static pj_bool_t on_rx_request( pjsip_rx_data *rdata ) { pj_sockaddr hostaddr; char temp[80], hostip[PJ_INET6_ADDRSTRLEN]; pj_str_t local_uri; pjsip_dialog *dlg; pjmedia_sdp_session *local_sdp; pjsip_tx_data *tdata; unsigned options = 0; pj_status_t status; /* * Respond (statelessly) any non-INVITE requests with 500 */ if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) { if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) { pj_str_t reason = pj_str("Simple UA unable to handle " "this request"); pjsip_endpt_respond_stateless( g_endpt, rdata, 500, &reason, NULL, NULL); } return PJ_TRUE; } /* * Reject INVITE if we already have an INVITE session in progress. */ if (g_inv) { pj_str_t reason = pj_str("Another call is in progress"); pjsip_endpt_respond_stateless( g_endpt, rdata, 500, &reason, NULL, NULL); return PJ_TRUE; } /* Verify that we can handle the request. */ status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, g_endpt, NULL); if (status != PJ_SUCCESS) { pj_str_t reason = pj_str("Sorry Simple UA can not handle this INVITE"); pjsip_endpt_respond_stateless( g_endpt, rdata, 500, &reason, NULL, NULL); return PJ_TRUE; } /* * Generate Contact URI */ if (pj_gethostip(AF, &hostaddr) != PJ_SUCCESS) { app_perror(THIS_FILE, "Unable to retrieve local host IP", status); return PJ_TRUE; } pj_sockaddr_print(&hostaddr, hostip, sizeof(hostip), 2); pj_ansi_sprintf(temp, "", hostip, SIP_PORT); local_uri = pj_str(temp); /* * Create UAS dialog. */ status = pjsip_dlg_create_uas( pjsip_ua_instance(), rdata, &local_uri, /* contact */ &dlg); if (status != PJ_SUCCESS) { pjsip_endpt_respond_stateless(g_endpt, rdata, 500, NULL, NULL, NULL); return PJ_TRUE; } /* * Get media capability from media endpoint: */ status = pjmedia_endpt_create_sdp( g_med_endpt, rdata->tp_info.pool, 1, &g_med_tpinfo.sock_info, &local_sdp); PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE); /* * Create invite session, and pass both the UAS dialog and the SDP * capability to the session. */ status = pjsip_inv_create_uas( dlg, rdata, local_sdp, 0, &g_inv); PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE); /* * Initially send 180 response. * * The very first response to an INVITE must be created with * pjsip_inv_initial_answer(). Subsequent responses to the same * transaction MUST use pjsip_inv_answer(). */ status = pjsip_inv_initial_answer(g_inv, rdata, 180, NULL, NULL, &tdata); PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE); /* Send the 180 response. */ status = pjsip_inv_send_msg(g_inv, tdata); PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE); /* * Now create 200 response. */ status = pjsip_inv_answer( g_inv, 200, NULL, /* st_code and st_text */ NULL, /* SDP already specified */ &tdata); PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE); /* * Send the 200 response. */ status = pjsip_inv_send_msg(g_inv, tdata); PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE); /* Done. * When the call is disconnected, it will be reported via the callback. */ return PJ_TRUE; } /* * Callback when SDP negotiation has completed. * We are interested with this callback because we want to start media * as soon as SDP negotiation is completed. */ static void call_on_media_update( pjsip_inv_session *inv, pj_status_t status) { pjmedia_session_info sess_info; const pjmedia_sdp_session *local_sdp; const pjmedia_sdp_session *remote_sdp; pjmedia_port *media_port; if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "SDP negotiation has failed", status); /* Here we should disconnect call if we're not in the middle * of initializing an UAS dialog and if this is not a re-INVITE. */ return; } /* Get local and remote SDP. * We need both SDPs to create a media session. */ status = pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp); status = pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp); /* Create session info based on the two SDPs. * We only support one stream per session for now. */ status = pjmedia_session_info_from_sdp(inv->dlg->pool, g_med_endpt, 1, &sess_info, local_sdp, remote_sdp); if (status != PJ_SUCCESS) { app_perror( THIS_FILE, "Unable to create media session", status); return; } /* If required, we can also change some settings in the session info, * (such as jitter buffer settings, codec settings, etc) before we * create the session. */ /* Create new media session, passing the two SDPs, and also the * media socket that we created earlier. * The media session is active immediately. */ status = pjmedia_session_create( g_med_endpt, &sess_info, &g_med_transport, NULL, &g_med_session ); if (status != PJ_SUCCESS) { app_perror( THIS_FILE, "Unable to create media session", status); return; } /* Get the media port interface of the first stream in the session. * Media port interface is basicly a struct containing get_frame() and * put_frame() function. With this media port interface, we can attach * the port interface to conference bridge, or directly to a sound * player/recorder device. */ pjmedia_session_get_port(g_med_session, 0, &media_port); /* Create a sound Player device and connect the media port to the * sound device. */ status = pjmedia_snd_port_create_player( inv->pool, /* pool */ -1, /* sound dev id */ media_port->info.clock_rate, /* clock rate */ media_port->info.channel_count, /* channel count */ media_port->info.samples_per_frame, /* samples per frame*/ media_port->info.bits_per_sample, /* bits per sample */ 0, /* options */ &g_snd_player); if (status != PJ_SUCCESS) { app_perror( THIS_FILE, "Unable to create sound player", status); PJ_LOG(3,(THIS_FILE, "%d %d %d %d", media_port->info.clock_rate, /* clock rate */ media_port->info.channel_count, /* channel count */ media_port->info.samples_per_frame, /* samples per frame*/ media_port->info.bits_per_sample /* bits per sample */ )); return; } status = pjmedia_snd_port_connect(g_snd_player, media_port); /* Create a sound recorder device and connect the media port to the * sound device. */ status = pjmedia_snd_port_create_rec( inv->pool, /* pool */ -1, /* sound dev id */ media_port->info.clock_rate, /* clock rate */ media_port->info.channel_count, /* channel count */ media_port->info.samples_per_frame, /* samples per frame*/ media_port->info.bits_per_sample, /* bits per sample */ 0, /* options */ &g_snd_rec); if (status != PJ_SUCCESS) { app_perror( THIS_FILE, "Unable to create sound recorder", status); return; } status = pjmedia_snd_port_connect(g_snd_rec, media_port); /* Done with media. */ }