本文介绍了如何利用pjsip2.5.5库中的samples工程，特别是`simpleua.c`文件，来创建一个无需注册到SIP服务器的基本UA。该UA支持基本呼叫、UDP在5060端口传输、RTP在4000端口传输、仅使用PCMA和PCMU编码器的语音媒体。通过调用`simpleua sip地址`命令，程序会进行SDP协商并启动RTP媒体传输。呼入呼叫会自动应答并挂断。代码示例展示了如何处理SIP请求和响应，以及媒体流的创建和管理。
这是pjsip2.5.5的samples工程内提供包括媒体及完整UA功能的简单应用，文件位置：pjproject-2.5.5\pjsip-apps\src\samples\simpleua.c，用户代理（UA）在SDK协商成功后启动RTP媒体传输。
此程序不需要注册到SIP服务器，它能够完成：基本呼叫、在5060端口传输UDP、在4000端口传输RTP、SDP协商、语音编解码器只支持PCMA和PCMU、声卡的语音媒体。
Samples工程编译完成后，在命令行输入：
simpleua 要呼叫的sip地址
要呼叫的sip地址格式： sip:用户名@对端IP
*呼入的呼叫先以180自动应答，再以200自动应答，此程序单次呼叫后即退出。
#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>
#define THIS_FILE "simpleua.c"
#include "util.h"
//设置
#define AF pj_AF_INET()// 如果使用IPV6，将此行改为pj_AF_INET6()
//必须设置PJ_HAS_IPV6
//并且操作系统也需要支持 IPv6. */
#if 0
#define SIP_PORT 5080 //侦听的SIP端口
#define RTP_PORT 5000 // RTP端口
#else
#define SIP_PORT 5060 //侦听的SIP端口
#define RTP_PORT 4000 // RTP端口
#endif
#define MAX_MEDIA_CNT 2 //媒体数量，设置为1将支持音频，
//设置为2，即支持音频，又支持视频
//全局静态变量
static pj_bool_t g_complete; //退出状态
static pjsip_endpoint *g_endpt; // SIP终端
static pj_caching_pool cp; //全局pool factory
static pjmedia_endpt *g_med_endpt; // 媒体终端
staticpjmedia_transport_info g_med_tpinfo[MAX_MEDIA_CNT];
//媒体传输用套接字信息
staticpjmedia_transport *g_med_transport[MAX_MEDIA_CNT];
//媒体流传输端口
staticpjmedia_sock_info g_sock_info[MAX_MEDIA_CNT];
//套接字信息组
//呼叫部分变量
staticpjsip_inv_session *g_inv; //当前invite传话
static pjmedia_stream *g_med_stream; //语音流
static pjmedia_snd_port *g_snd_port; //声卡设备
#if defined(PJMEDIA_HAS_VIDEO)&& (PJMEDIA_HAS_VIDEO != 0)
staticpjmedia_vid_stream *g_med_vstream; //视频流
static pjmedia_vid_port *g_vid_capturer;//视频捕获设备
static pjmedia_vid_port *g_vid_renderer;//视频播放设备
#endif //PJMEDIA_HAS_VIDEO
//函数声明
//呼叫过程上，当SDP协商完成后，回调此函数
static void call_on_media_update( pjsip_inv_session *inv,
pj_status_t status);
//当invite会话状态变化后，回调此函数
static void call_on_state_changed( pjsip_inv_session *inv,
pjsip_event *e);
//当新对话建立后，回调此函数
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
//对话之外，当收到请求时，回调此函数
static pj_bool_t on_rx_request( pjsip_rx_data *rdata );
//此PJSIP模块注册到应用用于处理对话或事务之外的请求，主要目的是处理INVITE请求消息，在那里将为它创建新的对话和INVITE会话
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() */
};
//呼入消息通知
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));
//此处必须返回false,否则其它消息将不被处理
return PJ_FALSE;
}
//呼出消息通知
static pj_status_t logging_on_tx_msg(pjsip_tx_data *tdata)
{
/* Important note:
* tp_infofield is only valid after outgoing messages has passed
* transportlayer. So don't try to access tp_info when the module
* haslower 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 notget sent! */
return PJ_SUCCESS;
}
//模块实例
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 asSIP 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;
/* 必须先初始化PJLIB*/
status = pj_init();
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
pj_log_set_level(5);
/* 再初始化PJLIB-UTIL*/
status = pjlib_util_init();
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
/* 在分配内存之前必须创建pool factory*/
pj_caching_pool_init(&cp,&pj_pool_factory_default_policy, 0);
//创建全局终端
{
const pj_str_t *hostname;
const char *endpt_name;
//终端必须分配全局唯一名称，此处简单地使用主机名实现
hostname = pj_gethostname();
endpt_name = hostname->ptr;
//创建终端
status = pjsip_endpt_create(&cp.factory,endpt_name,
&g_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
}
//添加UDP传输端口
//如果已经存在了UDP套接字时，应用可使用pjsip_udp_transport_attach()函数启动UDP传输端口
{
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, "Unableto start UDP transport", status);
return 1;
}
}
//初始化事务层，将创建/初始化传输hash表
status =pjsip_tsx_layer_init_module(g_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
//初始化UA层模块，将创建/初始化对话hash表
status = pjsip_ua_init_module( g_endpt, NULL );
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
//初始化invite会话模块，将初始化会话附加参数，例如与事件相关的回调函数
//on_state_changed与on_new_session是应用必须支持的回调函数
//我们可以让应用程序在on_media_update()回调函数中，启动媒体传输
{
pjsip_inv_callback inv_cb;
// 初始化INVITE会话回调
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;
//初始化INVITE会话模块
status = pjsip_inv_usage_init(g_endpt,&inv_cb);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
}
//初始化100rel支持
status = pjsip_100rel_init_module(g_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS,status);
//注册用于接收呼入请求的自己模块
status = pjsip_endpt_register_module(g_endpt, &mod_simpleua);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
//注册消息日志模块
status = pjsip_endpt_register_module(g_endpt, &msg_logger);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
//初始化媒体终端，它将隐性地启动PJMEDIA的相关功能
#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);
//给媒体终端增加PCMA/PCMU编解码器
#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)
//初始化视频子系统
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_OPENH264_CODEC) &&PJMEDIA_HAS_OPENH264_CODEC != 0
status =pjmedia_codec_openh264_vid_init(NULL, &cp.factory);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
# endif
# if defined(PJMEDIA_HAS_FFMPEG_VID_CODEC) &&PJMEDIA_HAS_FFMPEG_VID_CODEC!=0
//初始化ffmpeg视频编解码器
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 */
//创建用于RTP/RTCP套接字发送/接收的媒体传输端口
//每个呼叫均需要一个媒体传输端口，应用程序可以选择地复用相同的媒体传输端口用于后续呼叫
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, "Unableto create media transport", status);
return 1;
}
//取媒体传输端口的套接字信息(地址，端口)，我们需要利用它们创建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));
}
//如果提供呼叫的URL，则创建立即创建呼叫
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, "Unableto 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);
//创建UAC对话
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, "Unableto create UAC dialog", status);
return 1;
}
//如果怕呼出的INVITE被对方怀疑，我们可以在对话中加入凭证，如下例：
/*
{
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 outgoingINVITE, 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);
//创建INVITE传话，方便地将SDP作为初始参数传递给会话
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 tospecific 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.
* Toset 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!
*/
//创建INVITE初始请求
//此INVITE请求须包含完整的请求和SDP内容
status = pjsip_inv_invite(g_inv,&tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
//发送INVITE请求
//INVITE传话状态会通过回调传回
status = pjsip_inv_send_msg(g_inv, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
} else {
//没有外呼的URL
PJ_LOG(3,(THIS_FILE, "Ready toaccept incoming calls..."));
}
//循环，直到某个呼叫完成
for (;!g_complete;) {
pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(g_endpt,&timeout);
}
//退出时，转储当前内存使用情况
dump_pool_usage(THIS_FILE, &cp);
//销毁音频端口
//因为音频端口有线程存/取帧数据，故需要在流之前销毁音频端口
if (g_snd_port)
pjmedia_snd_port_destroy(g_snd_port);
#if defined(PJMEDIA_HAS_VIDEO)&& (PJMEDIA_HAS_VIDEO != 0)
//销毁视频端口
if (g_vid_capturer)
pjmedia_vid_port_destroy(g_vid_capturer);
if (g_vid_renderer)
pjmedia_vid_port_destroy(g_vid_renderer);
#endif
//销毁流
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);
//ffmpeg
# if defined(PJMEDIA_HAS_FFMPEG_VID_CODEC) &&PJMEDIA_HAS_FFMPEG_VID_CODEC!=0
pjmedia_codec_ffmpeg_vid_deinit();
# endif
# if defined(PJMEDIA_HAS_OPENH264_CODEC) &&PJMEDIA_HAS_OPENH264_CODEC != 0
pjmedia_codec_openh264_vid_deinit();
# endif
#endif
//销毁媒体传输端口
for (i = 0; i < MAX_MEDIA_CNT; ++i) {
if (g_med_transport[i])
pjmedia_transport_close(g_med_transport[i]);
}
//完结媒体终端
if (g_med_endpt)
pjmedia_endpt_destroy(g_med_endpt);
//完结sip终端
if (g_endpt)
pjsip_endpt_destroy(g_endpt);
//释放pool
if (pool)
pj_pool_release(pool);
return 0;
}
//当INVITE会话状态改变时的回调函数。
//此函数是在INVITE传话模块初始时注册，我们通常在此得到INVITE传话中断，并退出应用。
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, "CallDISCONNECTED [reason=%d (%s)]",
inv->cause,
pjsip_get_status_text(inv->cause)->ptr));
PJ_LOG(3,(THIS_FILE, "One callcompleted, application quitting..."));
g_complete = 1;
} else {
PJ_LOG(3,(THIS_FILE, "Call statechanged to %s",
pjsip_inv_state_name(inv->state)));
}
}
//当会话复制完成后，此函数被回调
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e)
{
/* To be done... */
PJ_UNUSED_ARG(inv);
PJ_UNUSED_ARG(e);
}
//当外部呼叫的任意对话或事务到达时，此函数被回调。我们可以控制想接的请求，其他的回应500拒绝
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;
//以500回应（无状态）任何非INVITE请求
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;
}
//如果此INVITE会话已经处理，则拒绝
if (g_inv) {
pj_str_t reason = pj_str("Another callis in progress");
pjsip_endpt_respond_stateless( g_endpt,rdata,
500, &reason,
NULL, NULL);
return PJ_TRUE;
}
//效验下我们要控制的请求
status = pjsip_inv_verify_request(rdata,&options, NULL, NULL,
g_endpt, NULL);
if (status != PJ_SUCCESS) {
pj_str_t reason = pj_str("Sorry SimpleUA can not handle this INVITE");
pjsip_endpt_respond_stateless( g_endpt,rdata,
500, &reason,
NULL, NULL);
return PJ_TRUE;
}
//生成联系URI
if (pj_gethostip(AF, &hostaddr) !=PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable toretrieve 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);
//创建UAS对话
status = pjsip_dlg_create_uas_and_inc_lock(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;
}
//从媒体终端得到媒体能力集
status = pjmedia_endpt_create_sdp( g_med_endpt,rdata->tp_info.pool,
MAX_MEDIA_CNT, g_sock_info,&local_sdp);
pj_assert(status == PJ_SUCCESS);
if (status != PJ_SUCCESS) {
pjsip_dlg_dec_lock(dlg);
return PJ_TRUE;
}
//传递UAS会话和SDP能力集，创建INVITE会话
status = pjsip_inv_create_uas( dlg, rdata,local_sdp, 0, &g_inv);
pj_assert(status == PJ_SUCCESS);
if (status != PJ_SUCCESS) {
pjsip_dlg_dec_lock(dlg);
return PJ_TRUE;
}
//INVITE传话已经创建，释放对话锁定
pjsip_dlg_dec_lock(dlg);
//首先发送180回应
//最先发送的回应必须用pjsip_inv_initial_answer()创建，后面相同事务的回应必须用pjsip_inv_answer()
status = pjsip_inv_initial_answer(g_inv,rdata,
180,
NULL, NULL, &tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS,PJ_TRUE);
//发送180回应
status = pjsip_inv_send_msg(g_inv, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS,PJ_TRUE);
//现在创建200回应
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);
//发送200回应
status = pjsip_inv_send_msg(g_inv, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS,PJ_TRUE);
//完成。当呼叫中断时，将在回调中收到报告
return PJ_TRUE;
}
//当SDP协商完成时，收到回调。我们要到此回调内快速启动媒体
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, "SDPnegotiation has failed", status);
/* Here we should disconnect call if we're not inthe middle
* ofinitializing an UAS dialog and if this is not a re-INVITE.
*/
return;
}
//取本地和远程SDP。我们需要这些SDP创建媒体会话
status =pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
status =pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
//SDP中的音频媒体描述创建流信息
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 tocreate audio stream info",status);
return;
}
//如果需要，在创建流之前，我们可以更改流信息中的设置，如jitter设置、编解码器设置等等。
//通过流信息和媒体套接字，轻松创建新的音频媒体流
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 tocreate audio stream", status);
return;
}
//启动音频流
status =pjmedia_stream_start(g_med_stream);
if (status != PJ_SUCCESS) {
app_perror( THIS_FILE, "Unable tostart audio stream", status);
return;
}
//取音频流的媒体端口接口。媒体端口接口基本结构包括get_frame()和put_frame()函数，
//我们可以将此媒体端口接口连接到会议桥，或直接到一个用于录音/放音的声卡设备
pjmedia_stream_get_port(g_med_stream,&media_port);
//创建声卡端口
pjmedia_snd_port_create(inv->pool,
PJMEDIA_AUD_DEFAULT_CAPTURE_DEV,
PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV,
PJMEDIA_PIA_SRATE(&media_port->info),//时间速率
PJMEDIA_PIA_CCNT(&media_port->info),//通话数
PJMEDIA_PIA_SPF(&media_port->info), //每帧采样
PJMEDIA_PIA_BITS(&media_port->info),//每次采样bit数
0,
&g_snd_port);
if (status != PJ_SUCCESS) {
app_perror( THIS_FILE, "Unable tocreate 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);
//取会话中第2个流的媒体端口接口，视频流的话，我们可以将此媒体端口接口直接连接到渲染/捕获视频设备
#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);
//通过SDP中的视频媒体描述合建视频信息
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;
}
//如果需要，在创建视频流之前，我们可以更改流信息中的设置，如jitter设置、编解码器设置等等。
//通过流信息和媒体套接字参数，创建新的视频媒体流
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;
}
//启动视频流
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 getdefault param of video "
"rendererdevice",status);
return;
}
//取解码的视频流
pjmedia_vid_stream_get_port(g_med_vstream, PJMEDIA_DIR_DECODING,
&media_port);
//设置格式
pjmedia_format_copy(&vport_param.vidparam.fmt,
&media_port->info.fmt);
vport_param.vidparam.dir = PJMEDIA_DIR_RENDER;
vport_param.active = PJ_TRUE;
//创建渲染端口
status = pjmedia_vid_port_create(inv->pool, &vport_param,
&g_vid_renderer);
if (status != PJ_SUCCESS){
app_perror(THIS_FILE, "Unable tocreate video renderer device",
status);
return;
}
//连接渲染端口到媒体端口
status = pjmedia_vid_port_connect(g_vid_renderer, media_port,
PJ_FALSE);
if (status !=PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable toconnect renderer to stream",
status);
return;
}
}
//创建捕获设备
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 getdefault param of video "
"capturedevice",status);
return;
}
//取编码的视频流端口
pjmedia_vid_stream_get_port(g_med_vstream, PJMEDIA_DIR_ENCODING,
&media_port);
//从流信息中取捕获格式
pjmedia_format_copy(&vport_param.vidparam.fmt,
&media_port->info.fmt);
vport_param.vidparam.dir = PJMEDIA_DIR_CAPTURE;
vport_param.active = PJ_TRUE;
//创建录音端口
status = pjmedia_vid_port_create(inv->pool, &vport_param,
&g_vid_capturer);
if (status !=PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable tocreate video capture device",
status);
return;
}
//连接录音媒体端口
status = pjmedia_vid_port_connect(g_vid_capturer, media_port,
PJ_FALSE);
if (status !=PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable toconnect capturer to stream",
status);
return;
}
}
//启动流
if (g_vid_renderer) {
status = pjmedia_vid_port_start(g_vid_renderer);
if (status !=PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable tostart 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 tostart video capturer",
status);
return;
}
}
}
#endif /* PJMEDIA_HAS_VIDEO */
//媒体部分完成
}
扫一扫
315
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
