| /* $Id$ */ |
| /* |
| * 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 |
| */ |
| |
| |
| /** |
| * 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 <pjsip.h> |
| #include <pjmedia.h> |
| #include <pjmedia-codec.h> |
| #include <pjsip_ua.h> |
| #include <pjsip_simple.h> |
| #include <pjlib-util.h> |
| #include <pjlib.h> |
| |
| /* 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, "<sip:simpleuac@%s:%d>", |
| 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].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, "<sip:simpleuas@%s:%d>", |
| 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. */ |
| } |
| |
| |