* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/pjsip-apps/src/samples/simpleua.c b/jni/pjproject-android/pjsip-apps/src/samples/simpleua.c
new file mode 100644
index 0000000..2906323
--- /dev/null
+++ b/jni/pjproject-android/pjsip-apps/src/samples/simpleua.c
@@ -0,0 +1,1030 @@
+/* $Id: simpleua.c 4051 2012-04-13 08:16:30Z ming $ */
+/*
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+/**
+ * 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. */
+#if 0
+#define SIP_PORT 5080 /* Listening SIP port */
+#define RTP_PORT 5000 /* RTP port */
+#else
+#define SIP_PORT 5060 /* Listening SIP port */
+#define RTP_PORT 4000 /* RTP port */
+#endif
+
+#define MAX_MEDIA_CNT 2 /* Media count, set to 1 for audio
+ * only or 2 for audio and video */
+
+/*
+ * 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[MAX_MEDIA_CNT];
+ /* Socket info for media */
+static pjmedia_transport *g_med_transport[MAX_MEDIA_CNT];
+ /* Media stream transport */
+static pjmedia_sock_info g_sock_info[MAX_MEDIA_CNT];
+ /* Socket info array */
+
+/* Call variables: */
+static pjsip_inv_session *g_inv; /* Current invite session. */
+static pjmedia_stream *g_med_stream; /* Call's audio stream. */
+static pjmedia_snd_port *g_snd_port; /* Sound device. */
+
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+static pjmedia_vid_stream *g_med_vstream; /* Call's video stream. */
+static pjmedia_vid_port *g_vid_capturer;/* Call's video capturer. */
+static pjmedia_vid_port *g_vid_renderer;/* Call's video renderer. */
+#endif /* PJMEDIA_HAS_VIDEO */
+
+/*
+ * 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() */
+};
+
+
+/* Notification on incoming messages */
+static pj_bool_t logging_on_rx_msg(pjsip_rx_data *rdata)
+{
+ PJ_LOG(4,(THIS_FILE, "RX %d bytes %s from %s %s:%d:\n"
+ "%.*s\n"
+ "--end msg--",
+ rdata->msg_info.len,
+ pjsip_rx_data_get_info(rdata),
+ rdata->tp_info.transport->type_name,
+ rdata->pkt_info.src_name,
+ rdata->pkt_info.src_port,
+ (int)rdata->msg_info.len,
+ 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 logging_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 %s:%d:\n"
+ "%.*s\n"
+ "--end msg--",
+ (tdata->buf.cur - tdata->buf.start),
+ pjsip_tx_data_get_info(tdata),
+ tdata->tp_info.transport->type_name,
+ tdata->tp_info.dst_name,
+ tdata->tp_info.dst_port,
+ (int)(tdata->buf.cur - tdata->buf.start),
+ 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-msg-log", 13 }, /* Name. */
+ -1, /* Id */
+ PJSIP_MOD_PRIORITY_TRANSPORT_LAYER-1,/* Priority */
+ NULL, /* load() */
+ NULL, /* start() */
+ NULL, /* stop() */
+ NULL, /* unload() */
+ &logging_on_rx_msg, /* on_rx_request() */
+ &logging_on_rx_msg, /* on_rx_response() */
+ &logging_on_tx_msg, /* on_tx_request. */
+ &logging_on_tx_msg, /* 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_pool_t *pool = NULL;
+ pj_status_t status;
+ unsigned i;
+
+ /* Must init PJLIB first: */
+ status = pj_init();
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ pj_log_set_level(5);
+
+ /* 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);
+
+ /*
+ * Register message logger module.
+ */
+ status = pjsip_endpt_register_module( g_endpt, &msg_logger);
+ 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
+
+
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ /* Init video subsystem */
+ pool = pjmedia_endpt_create_pool(g_med_endpt, "Video subsystem", 512, 512);
+ status = pjmedia_video_format_mgr_create(pool, 64, 0, NULL);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjmedia_converter_mgr_create(pool, NULL);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjmedia_vid_codec_mgr_create(pool, NULL);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjmedia_vid_dev_subsys_init(&cp.factory);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+# if defined(PJMEDIA_HAS_FFMPEG_VID_CODEC) && PJMEDIA_HAS_FFMPEG_VID_CODEC!=0
+ /* Init ffmpeg video codecs */
+ status = pjmedia_codec_ffmpeg_vid_init(NULL, &cp.factory);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+# endif /* PJMEDIA_HAS_FFMPEG_VID_CODEC */
+
+#endif /* PJMEDIA_HAS_VIDEO */
+
+ /*
+ * 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.
+ */
+ for (i = 0; i < PJ_ARRAY_SIZE(g_med_transport); ++i) {
+ status = pjmedia_transport_udp_create3(g_med_endpt, AF, NULL, NULL,
+ RTP_PORT + i*2, 0,
+ &g_med_transport[i]);
+ 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[i]);
+ pjmedia_transport_get_info(g_med_transport[i], &g_med_tpinfo[i]);
+
+ pj_memcpy(&g_sock_info[i], &g_med_tpinfo[i].sock_info,
+ sizeof(pjmedia_sock_info));
+ }
+
+ /*
+ * 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].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.
+ */
+ status = pjmedia_endpt_create_sdp( g_med_endpt, /* the media endpt */
+ dlg->pool, /* pool. */
+ MAX_MEDIA_CNT, /* # of streams */
+ g_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);
+
+ /* Destroy audio ports. Destroy the audio port first
+ * before the stream since the audio port has threads
+ * that get/put frames to the stream.
+ */
+ if (g_snd_port)
+ pjmedia_snd_port_destroy(g_snd_port);
+
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ /* Destroy video ports */
+ if (g_vid_capturer)
+ pjmedia_vid_port_destroy(g_vid_capturer);
+ if (g_vid_renderer)
+ pjmedia_vid_port_destroy(g_vid_renderer);
+#endif
+
+ /* Destroy streams */
+ if (g_med_stream)
+ pjmedia_stream_destroy(g_med_stream);
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ if (g_med_vstream)
+ pjmedia_vid_stream_destroy(g_med_vstream);
+
+ /* Deinit ffmpeg codec */
+# if defined(PJMEDIA_HAS_FFMPEG_VID_CODEC) && PJMEDIA_HAS_FFMPEG_VID_CODEC!=0
+ pjmedia_codec_ffmpeg_vid_deinit();
+# endif
+
+#endif
+
+ /* Destroy media transports */
+ for (i = 0; i < MAX_MEDIA_CNT; ++i) {
+ if (g_med_transport[i])
+ pjmedia_transport_close(g_med_transport[i]);
+ }
+
+ /* Deinit pjmedia endpoint */
+ if (g_med_endpt)
+ pjmedia_endpt_destroy(g_med_endpt);
+
+ /* Deinit pjsip endpoint */
+ if (g_endpt)
+ pjsip_endpt_destroy(g_endpt);
+
+ /* Release pool */
+ if (pool)
+ pj_pool_release(pool);
+
+ 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,
+ MAX_MEDIA_CNT, g_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_stream_info stream_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 stream info based on the media audio SDP. */
+ status = pjmedia_stream_info_from_sdp(&stream_info, inv->dlg->pool,
+ g_med_endpt,
+ local_sdp, remote_sdp, 0);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE,"Unable to create audio stream info",status);
+ return;
+ }
+
+ /* If required, we can also change some settings in the stream info,
+ * (such as jitter buffer settings, codec settings, etc) before we
+ * create the stream.
+ */
+
+ /* Create new audio media stream, passing the stream info, and also the
+ * media socket that we created earlier.
+ */
+ status = pjmedia_stream_create(g_med_endpt, inv->dlg->pool, &stream_info,
+ g_med_transport[0], NULL, &g_med_stream);
+ if (status != PJ_SUCCESS) {
+ app_perror( THIS_FILE, "Unable to create audio stream", status);
+ return;
+ }
+
+ /* Start the audio stream */
+ status = pjmedia_stream_start(g_med_stream);
+ if (status != PJ_SUCCESS) {
+ app_perror( THIS_FILE, "Unable to start audio stream", status);
+ return;
+ }
+
+ /* Get the media port interface of the audio stream.
+ * 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_stream_get_port(g_med_stream, &media_port);
+
+ /* Create sound port */
+ pjmedia_snd_port_create(inv->pool,
+ PJMEDIA_AUD_DEFAULT_CAPTURE_DEV,
+ PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV,
+ PJMEDIA_PIA_SRATE(&media_port->info),/* clock rate */
+ PJMEDIA_PIA_CCNT(&media_port->info),/* channel count */
+ PJMEDIA_PIA_SPF(&media_port->info), /* samples per frame*/
+ PJMEDIA_PIA_BITS(&media_port->info),/* bits per sample */
+ 0,
+ &g_snd_port);
+
+ if (status != PJ_SUCCESS) {
+ app_perror( THIS_FILE, "Unable to create sound port", status);
+ PJ_LOG(3,(THIS_FILE, "%d %d %d %d",
+ PJMEDIA_PIA_SRATE(&media_port->info),/* clock rate */
+ PJMEDIA_PIA_CCNT(&media_port->info),/* channel count */
+ PJMEDIA_PIA_SPF(&media_port->info), /* samples per frame*/
+ PJMEDIA_PIA_BITS(&media_port->info) /* bits per sample */
+ ));
+ return;
+ }
+
+ status = pjmedia_snd_port_connect(g_snd_port, media_port);
+
+
+ /* Get the media port interface of the second stream in the session,
+ * which is video stream. With this media port interface, we can attach
+ * the port directly to a renderer/capture video device.
+ */
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ if (local_sdp->media_count > 1) {
+ pjmedia_vid_stream_info vstream_info;
+ pjmedia_vid_port_param vport_param;
+
+ pjmedia_vid_port_param_default(&vport_param);
+
+ /* Create stream info based on the media video SDP. */
+ status = pjmedia_vid_stream_info_from_sdp(&vstream_info,
+ inv->dlg->pool, g_med_endpt,
+ local_sdp, remote_sdp, 1);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE,"Unable to create video stream info",status);
+ return;
+ }
+
+ /* If required, we can also change some settings in the stream info,
+ * (such as jitter buffer settings, codec settings, etc) before we
+ * create the video stream.
+ */
+
+ /* Create new video media stream, passing the stream info, and also the
+ * media socket that we created earlier.
+ */
+ status = pjmedia_vid_stream_create(g_med_endpt, NULL, &vstream_info,
+ g_med_transport[1], NULL,
+ &g_med_vstream);
+ if (status != PJ_SUCCESS) {
+ app_perror( THIS_FILE, "Unable to create video stream", status);
+ return;
+ }
+
+ /* Start the video stream */
+ status = pjmedia_vid_stream_start(g_med_vstream);
+ if (status != PJ_SUCCESS) {
+ app_perror( THIS_FILE, "Unable to start video stream", status);
+ return;
+ }
+
+ if (vstream_info.dir & PJMEDIA_DIR_DECODING) {
+ status = pjmedia_vid_dev_default_param(
+ inv->pool, PJMEDIA_VID_DEFAULT_RENDER_DEV,
+ &vport_param.vidparam);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to get default param of video "
+ "renderer device", status);
+ return;
+ }
+
+ /* Get video stream port for decoding direction */
+ pjmedia_vid_stream_get_port(g_med_vstream, PJMEDIA_DIR_DECODING,
+ &media_port);
+
+ /* Set format */
+ pjmedia_format_copy(&vport_param.vidparam.fmt,
+ &media_port->info.fmt);
+ vport_param.vidparam.dir = PJMEDIA_DIR_RENDER;
+ vport_param.active = PJ_TRUE;
+
+ /* Create renderer */
+ status = pjmedia_vid_port_create(inv->pool, &vport_param,
+ &g_vid_renderer);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to create video renderer device",
+ status);
+ return;
+ }
+
+ /* Connect renderer to media_port */
+ status = pjmedia_vid_port_connect(g_vid_renderer, media_port,
+ PJ_FALSE);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to connect renderer to stream",
+ status);
+ return;
+ }
+ }
+
+ /* Create capturer */
+ if (vstream_info.dir & PJMEDIA_DIR_ENCODING) {
+ status = pjmedia_vid_dev_default_param(
+ inv->pool, PJMEDIA_VID_DEFAULT_CAPTURE_DEV,
+ &vport_param.vidparam);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to get default param of video "
+ "capture device", status);
+ return;
+ }
+
+ /* Get video stream port for decoding direction */
+ pjmedia_vid_stream_get_port(g_med_vstream, PJMEDIA_DIR_ENCODING,
+ &media_port);
+
+ /* Get capturer format from stream info */
+ pjmedia_format_copy(&vport_param.vidparam.fmt,
+ &media_port->info.fmt);
+ vport_param.vidparam.dir = PJMEDIA_DIR_CAPTURE;
+ vport_param.active = PJ_TRUE;
+
+ /* Create capturer */
+ status = pjmedia_vid_port_create(inv->pool, &vport_param,
+ &g_vid_capturer);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to create video capture device",
+ status);
+ return;
+ }
+
+ /* Connect capturer to media_port */
+ status = pjmedia_vid_port_connect(g_vid_capturer, media_port,
+ PJ_FALSE);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to connect capturer to stream",
+ status);
+ return;
+ }
+ }
+
+ /* Start streaming */
+ if (g_vid_renderer) {
+ status = pjmedia_vid_port_start(g_vid_renderer);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to start video renderer",
+ status);
+ return;
+ }
+ }
+ if (g_vid_capturer) {
+ status = pjmedia_vid_port_start(g_vid_capturer);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to start video capturer",
+ status);
+ return;
+ }
+ }
+ }
+#endif /* PJMEDIA_HAS_VIDEO */
+
+ /* Done with media. */
+}
+
+