环境搭建等参考：JRTP实时音视频传输(1)-必做的环境搭建与demo测试
1.创建自己的demo
先将example1拷贝为myclienttcp.cpp和myservertcp.cpp
cp example1.cpp myclienttcp.cpp
cp example1.cpp myservertcp.cpp
改写jrtplib/JRTPLIB/examples/CMakeLists.txt，添加myclienttcp和myservertcp编译
重新生成Makefile并编译
shell
sudo cmake CMakeLists.txt
sudo make可以看到成功编译了myclienttcp和myservertcp源文件
编译通过，这里就去实现demo就行
2.demo源码-客户端
cpp
#include <iostream>
#include <arpa/inet.h>
#include "rtptcpaddress.h"
#include "rtpsession.h"
#include "rtpsessionparams.h"
#include "rtptcptransmitter.h"
#include "rtpipv4address.h"
#include "rtptimeutilities.h"
#include "rtppacket.h"
#include "rtpabortdescriptors.h"
using namespace jrtplib;
#define SERVER_IP "127.0.0.1"
#define SERVER_PORT 58008
int main()
{
RTPSession session;
RTPAbortDescriptors m_descriptors;
RTPSessionParams sessionparams;
sessionparams.SetAcceptOwnPackets(true);
sessionparams.SetOwnTimestampUnit(1.0/10.0);
m_descriptors.Init();
RTPTCPTransmissionParams transparams;
transparams.SetCreatedAbortDescriptors(&m_descriptors);
int status = session.Create(sessionparams,&transparams,RTPTransmitter::TCPProto);
if (status < 0)
{
printf("my client session create failed\n");
return -1;
}
//初始化socket
int sock = socket(AF_INET, SOCK_STREAM, 0);
sockaddr_in addrSrv;
addrSrv.sin_addr.s_addr = inet_addr(SERVER_IP);
addrSrv.sin_family = AF_INET;
addrSrv.sin_port = htons(SERVER_PORT);
printf("my client prepare to connect\n");
//连接服务器
connect( sock, (sockaddr*)&addrSrv, sizeof(sockaddr));
RTPTCPAddress addr(sock);
status = session.AddDestination(addr);
if (status < 0)
{
printf("my client session add destination failed\n");
return -1;
}
session.SetDefaultPayloadType(96);
session.SetDefaultMark(false);
session.SetDefaultTimestampIncrement(160);
for (int i = 0; i < 50 ; i++)
{
std::string str("123456");
//发送数据
session.SendPacket((void *)str.c_str(), str.length(),0,false,10);
printf("my client send packet:%s, len:%d, idx:%d\n", str.c_str(), str.length(), i);
RTPTime::Wait(RTPTime(10, 0));
}
RTPTime delay(0.020);
session.BYEDestroy(delay,"Client End",9);
}3.demo源码-服务端
cpp
/*
Here's a small IPv4 example: it asks for a portbase and a destination and
starts sending packets to that destination.
*/
#include <sys/types.h>
#include <sys/socket.h>
#include "rtppacket.h"
#include "rtptcpaddress.h"
#include "rtptcptransmitter.h"
#include "rtpsession.h"
#include "rtpudpv4transmitter.h"
#include "rtpipv4address.h"
#include "rtpsessionparams.h"
#include "rtperrors.h"
#include "rtplibraryversion.h"
#include <stdlib.h>
#include <stdio.h>
#include <iostream>
#include <string>
using namespace jrtplib;
#define SERVER_PORT 58008
void checkerror(int rtperr)
{
if (rtperr < 0)
{
std::cout << "ERROR: " << RTPGetErrorString(rtperr) << std::endl;
exit(-1);
}
}
int main(void)
{
int nListener = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
if (nListener == -1)
{
return -1;
}
sockaddr_in serverAddr;
memset(&serverAddr, 0, sizeof(sockaddr_in));
serverAddr.sin_family = AF_INET;
serverAddr.sin_addr.s_addr = INADDR_ANY;
serverAddr.sin_port = htons(SERVER_PORT);
int nRet = bind(nListener, (sockaddr*)&serverAddr, sizeof(serverAddr));
if (nRet == -1)
{
return -1;
}
if (listen(nListener, 1) == -1)
{
return -1;
}
printf("my server is listen ready, wait for connect\n");
sockaddr_in clientAddr;
int nLen = sizeof(sockaddr_in);
int nServer = -1;
while (true)
{
nServer = accept(nListener, (sockaddr*)&clientAddr, (socklen_t *)&nLen);
if (nServer == -1)
{
continue;
}
else
{
break;
}
}
printf("my server connect new client\n");
int status = -1;
int nPackSize = 45678;
RTPSessionParams sessparams;
RTPSession m_RTPTCPSession;
sessparams.SetProbationType(RTPSources::NoProbation);
sessparams.SetOwnTimestampUnit(90000.0 / 25.0);
sessparams.SetMaximumPacketSize(nPackSize + 64);
RTPTCPTransmitter *pTransparams = new RTPTCPTransmitter(NULL);
status = pTransparams->Init(false);
if (status < 0)
{
printf("my server trans param init failed, reason:%s\n", RTPGetErrorString(status).c_str());
return -1;
}
status = pTransparams->Create(65535, NULL);
if (status < 0)
{
printf("my server trans param create failed, reason:%s\n", RTPGetErrorString(status).c_str());
return -1;
}
status = m_RTPTCPSession.Create(sessparams, pTransparams);
if (status < 0)
{
printf("my server trans session create failed, reason:%s\n", RTPGetErrorString(status).c_str());
return -1;
}
status = m_RTPTCPSession.AddDestination(RTPTCPAddress(nServer));
if (status < 0)
{
printf("my server trans session add failed, reason:%s\n", RTPGetErrorString(status).c_str());
return -1;
}
while (1)
{
m_RTPTCPSession.BeginDataAccess();
// check incoming packets
if (m_RTPTCPSession.GotoFirstSourceWithData())
{
do
{
RTPPacket *pack;
while ((pack = m_RTPTCPSession.GetNextPacket()) != NULL)
{
// You can examine the data here
printf("myserver recv packet buf:%s, len:%d\n", pack->GetPayloadData(), pack->GetPayloadLength());
// we don't longer need the packet, so
// we'll delete it
m_RTPTCPSession.DeletePacket(pack);
}
} while (m_RTPTCPSession.GotoNextSourceWithData());
}
m_RTPTCPSession.EndDataAccess();
#ifndef RTP_SUPPORT_THREAD
status = m_RTPTCPSession.Poll();
checkerror(status);
#endif // RTP_SUPPORT_THREAD
RTPTime::Wait(RTPTime(1,0));
}
m_RTPTCPSession.BYEDestroy(RTPTime(10,0),0,0);
return 0;
}4.demo运行测试
分别运行client和server ，可以看到数据正常传输到server端
用netstat查看连接端口信息，也能看到该端口目前的状态，属于TCP连接，实验成功
对环境搭建不清楚的可以看这篇博客~
JRTP实时音视频传输(1)-必做的环境搭建与demo测试
5.源码下载
哈喽~我是Embedded-Xin，沪漂嵌入式开发工程师一枚，立志成为嵌入式全栈开发工程师，成为优秀博客创作者，共同学习进步。
以上代码全部放在我私人的github地址，其中有许多自己辛苦敲的例程源码，供大家参考、批评指正，有兴趣还可以直接提patch修改我的仓库~：
https://github.com/Xuzhangxin/study_linux_project.git
觉得不错的话可以点个收藏和star~