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simpleua.c
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/* $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
/* Comment out one of the following 3 macros.
*
* PLAYBACK_ONLY - play audio from file instead of using microphone
* RECORD_ONLY - record audio to file instead of using speakers
* PLAYBACK_AND_RECORD - play and record to files
*
* IF ALL OPTIONS ARE DISABLED the program behaves like original simpleua.c */
#define PLAYBACK_ONLY
//#define RECORD_ONLY
//#define PLAYBACK_AND_RECORD
#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;
}
struct play_rec_port {
pjmedia_port *rec_pj_port;
pjmedia_port *play_pj_port;
};
static struct play_rec_port global_rec_play_port;
static pjmedia_port global_pj_port;
static pj_status_t rec_play_port_get_frame(pjmedia_port *port,
pjmedia_frame *frame) {
pjmedia_port *p = global_rec_play_port.play_pj_port;
return p->get_frame(p, frame);
}
static pj_status_t rec_play_port_put_frame(pjmedia_port *port,
pjmedia_frame *frame) {
pjmedia_port *p = global_rec_play_port.rec_pj_port;
return p->put_frame(p, frame);
}
static void create_player_and_recorder(pjsip_inv_session *inv, pjmedia_port *media_port) {
pjmedia_master_port *master_port = NULL;
unsigned wav_ptime;
pj_status_t status;;
char play_file[] = "iron_man.wav";
char rec_file[] = "output.wav";
pjmedia_port *rec_file_port;
pjmedia_port *play_file_port;
wav_ptime = PJMEDIA_PIA_PTIME(&media_port->info);
status = pjmedia_wav_player_port_create(inv->pool, play_file, wav_ptime,
0, -1, &play_file_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to use play file", status);
return;
}
status = pjmedia_wav_writer_port_create(inv->pool,
rec_file,
PJMEDIA_PIA_SRATE(&media_port->info),
PJMEDIA_PIA_CCNT(&media_port->info),
PJMEDIA_PIA_SPF(&media_port->info),
PJMEDIA_PIA_BITS(&media_port->info),
0, 0,
&rec_file_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to use rec file", status);
return;
}
memcpy(&global_pj_port, rec_file_port, sizeof (pjmedia_port));
status = pjmedia_master_port_create(inv->pool,
media_port,
&global_pj_port,
0,
&master_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to create rec master port", status);
return;
}
global_rec_play_port.rec_pj_port = rec_file_port;
global_rec_play_port.play_pj_port = play_file_port;
global_pj_port.get_frame = rec_play_port_get_frame;
global_pj_port.put_frame = rec_play_port_put_frame;
status = pjmedia_master_port_start(master_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error starting play master port", status);
return;
}
printf("\n\n Playing from WAV %s and recording to WAV %s \n\n",
play_file, rec_file);
}
static void create_player(pjsip_inv_session *inv, pjmedia_port *media_port) {
unsigned wav_ptime;
pj_status_t status;;
char play_file[] = "iron_man.wav";
pjmedia_port *play_file_port = NULL;
pjmedia_master_port *master_port = NULL;
wav_ptime = PJMEDIA_PIA_PTIME(&media_port->info);
status = pjmedia_wav_player_port_create(inv->dlg->pool, play_file, wav_ptime,
0, -1, &play_file_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to use file", status);
return;
}
status = pjmedia_master_port_create(inv->dlg->pool, play_file_port, media_port,
0, &master_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to create master port", status);
return;
}
status = pjmedia_master_port_start(master_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error starting master port", status);
return;
}
printf("\n\nPlaying from WAV file %s..\n\n", play_file);
}
static void create_recorder(pjsip_inv_session *inv, pjmedia_port *media_port) {
pj_status_t status;;
char rec_file[] = "output.wav";
pjmedia_port *rec_file_port = NULL;
pjmedia_master_port *master_port = NULL;
status = pjmedia_wav_writer_port_create(inv->pool,
rec_file,
PJMEDIA_PIA_SRATE(&media_port->info),
PJMEDIA_PIA_CCNT(&media_port->info),
PJMEDIA_PIA_SPF(&media_port->info),
PJMEDIA_PIA_BITS(&media_port->info),
0, 0,
&rec_file_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to use rec file", status);
return;
}
status = pjmedia_master_port_create(inv->dlg->pool, media_port,
rec_file_port, 0, &master_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to create master port", status);
return;
}
status = pjmedia_master_port_start(master_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error starting master port", status);
return;
}
printf("\n\nRecording to WAV file %s..\n\n", rec_file);
}
/*
* 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;
}