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🤣本文内容🤣：🍭TCP传输H264格式RTP包(RTP_OVER_TCP)的RTSP服务器🍭
⏰发布时间⏰： 2025-07-16
本文未经允许，不得转发！！！
目录
前面系列文章回顾：
【音视频 | RTSP】RTSP协议详解 及 抓包例子解析（详细而不赘述）
【音视频 | RTSP】SDP(会话描述协议)详解 及 抓包例子分析
【音视频 | RTP】RTP协议详解(H.264的RTP封包格式、AAC的RTP封包格式)
【RTSP从零实践】01、根据RTSP协议实现一个RTSP服务
【RTSP从零实践】02、使用RTP协议封装并传输H264
【RTSP从零实践】03、实现最简单的传输H264的RTSP服务器
【RTSP从零实践】04、使用RTP协议封装并传输AAC
【RTSP从零实践】05、实现最简单的传输AAC的RTSP服务器
【RTSP从零实践】06、实现最简单的同时传输H264、AAC的RTSP服务器
【RTSP从零实践】07、多播传输H264格式的RTP包(附带源码)
【RTSP从零实践】08、多播传输H264码流的RTSP服务器——最简单的实现例子(附带源码)
【RTSP从零实践】09、多播传输AAC格式的RTP包(附带源码)
【RTSP从零实践】10、多播传输AAC码流的RTSP服务器——最简单的实现例子(附带源码)
【RTSP从零实践】11、多播同时传输H264、AAC码流的RTSP服务器——最简单的实现例子(附带源码)
【RTSP从零实践】12、TCP传输H264格式RTP包(RTP_over_TCP)的RTSP服务器(附带源码)
🎄一、概述
上篇文章介绍了使用TCP协议传输H264格式的RTP包，这篇介绍的是使用TCP协议传输AAC格式的RTP包，下面主要介绍 RTP over TCP(RTSP) 的相关概念，然后直接看代码。
🎄二、RTP over TCP(RTSP) 介绍
✨2.1 RTP over TCP(RTSP) 相关概念
我们前面系列文章介绍过的rtsp服务器都是创建了一个TCP服务来处理RTSP协议，创建另一个UDP套接字来发送RTP包，这种方式就是 RTP over UDP。而RTP over TCP不单独建立一个UDP套接字去发送RTP包，而是利用处理RTSP协议的TCP套接字来发送RTP包的，所有有些资料也把这种方式称为RTP over RTSP。
RTP over UDP：一个TCP套接字处理RTSP协议，另一个UDP套接字发送RTP包、RTCP包；RTP over TCP(RTSP)：一个TCP套接字处理RTSP协议、RTP包、RTCP包。
✨2.2 怎么区分 RTSP包、RTP包、RTCP包
如上面所说，我们复用发送RTSP交互的socket来发送RTP包和RTCP信息，那么对于客户端来说，如何区分这三种数据呢？
我们将这三个分为两类，一类是RTSP，一类是RTP、RTCP
发送RTSP信息的情况没有变化，还是更以前一样的方式
发送RTP、RTCP包，在每个包前面都加上四个字节，这四个字节解释如下：
| 字节 | 描述 |
|---|---|
| 第一个字节 | 字符'$'，表示这个包是RTP包 或 RTCP包 |
| 第二个字节 | 通道号channel，用于区分RTP包 或 RTCP包 |
| 第三、四个字节 | 表示RTP包的大小 |
使用TCP协议传输的RTP包和RTCP包，第一个字节固定为$字符，第二字节的channel是在RTSP服务器处理的SETUP过程中，客户端发送给服务端的。
所以，使用TCP协议传输的RTP包结构如下：
✨2.3 RTP包封装代码
在RTP包起始位置增加4个字节的数据：
struct RtpPacket
{
char header[4];
struct RtpHeader rtpHeader;
uint8_t payload[0];
};
在发送RTP包前，是这样填写这4个字节：
rtpPacket->header[0] = '$';
rtpPacket->header[1] = rtpChannel;
rtpPacket->header[2] = ((dataSize+RTP_HEADER_SIZE) & 0xFF00 ) >> 8;
rtpPacket->header[3] = (dataSize+RTP_HEADER_SIZE) & 0xFF;
🎄三、RTP_over_TCP的RTSP服务端源码
1、aacReader.h
/**
* @file aacReader.h
* @author : https://blog.csdn.net/wkd_007
* @brief
* @version 0.1
* @date 2025-06-30
*
* @copyright Copyright (c) 2025
*
*/
#ifndef __AAC_READER_H__
#define __AAC_READER_H__
#include <stdio.h>
#define ADTS_HEADER_LEN (7)
typedef struct
{
int frame_len; //!
unsigned char *pFrameBuf; //!
} AACFrame_t;
typedef struct AACReaderInfo_s
{
FILE *pFileFd;
}AACReaderInfo_t;
int AAC_FileOpen(char *fileName, AACReaderInfo_t *pAACInfo);
int AAC_FileClose(AACReaderInfo_t *pAACInfo);
int AAC_GetADTSFrame(AACFrame_t *pAACFrame, const AACReaderInfo_t *pAACInfo);
int AAC_IsEndOfFile(const AACReaderInfo_t *pAACInfo);
void AAC_SeekFile(const AACReaderInfo_t *pAACInfo);
#endif // __AAC_READER_H__
2、aacReader.c
/**
* @file aacReader.c
* @author : https://blog.csdn.net/wkd_007
* @brief
* @version 0.1
* @date 2025-06-30
*
* @copyright Copyright (c) 2025
*
*/
#include <stdlib.h>
#include <string.h>
#include "aacReader.h"
#define MAX_FRAME_LEN (1024*1024) // Ò»Ö¡Êý¾Ý×î´ó×Ö½ÚÊý
#define MAX_SYNCCODE_LEN (3) // ͬ²½Âë×Ö½Ú¸öÊý 2025-05-21 17:45:06
static int findSyncCode_0xFFF(unsigned char *Buf, int *size)
{
if((Buf[0] == 0xff) && ((Buf[1] & 0xf0) == 0xf0) )//0xFF F£¬Ç°12bit¶¼Îª1 2025-05-21 17:46:57
{
*size |= ((Buf[3] & 0x03) <<11); //high 2 bit
*size |= Buf[4]<<3; //middle 8 bit
*size |= ((Buf[5] & 0xe0)>>5); //low 3bit
return 1;
}
return 0;
}
int AAC_FileOpen(char *fileName, AACReaderInfo_t *pAACInfo)
{
pAACInfo->pFileFd = fopen(fileName, "rb+");
if (pAACInfo->pFileFd==NULL){
printf("[%s %d]Open file error\n",__FILE__,__LINE__);
return -1;
}
return 0;
}
int AAC_FileClose(AACReaderInfo_t *pAACInfo)
{
if (pAACInfo->pFileFd != NULL) {
fclose(pAACInfo->pFileFd);
pAACInfo->pFileFd = NULL;
}
return 0;
}
int AAC_IsEndOfFile(const AACReaderInfo_t *pAACInfo)
{
return feof(pAACInfo->pFileFd);
}
void AAC_SeekFile(const AACReaderInfo_t *pAACInfo)
{
fseek(pAACInfo->pFileFd,0,SEEK_SET);
}
/**
* @brief
*
* @param pAACFrame :Êä³ö²ÎÊý£¬Ê¹Óúó pAACInfo->pFrameBuf ÐèÒªfree
* @param pAACInfo
* @return int
*/
int AAC_GetADTSFrame(AACFrame_t *pAACFrame, const AACReaderInfo_t *pAACInfo)
{
int rewind = 0;
if (pAACInfo->pFileFd==NULL){
printf("[%s %d]pFileFd error\n",__FILE__,__LINE__);
return -1;
}
// 1.ÏȶÁÈ¡ADTSÖ¡Í·(7¸ö×Ö½Ú)
unsigned char* pFrame = (unsigned char*)malloc(MAX_FRAME_LEN);
int readLen = fread(pFrame, 1, ADTS_HEADER_LEN, pAACInfo->pFileFd);
if(readLen <= 0)
{
printf("[%s %d]fread error readLen=%d\n",__FILE__,__LINE__,readLen);
free(pFrame);
return -1;
}
// 2.²éÕÒµ±Ç°Ö¡Í¬²½Â룬»ñÈ¡Ö¡³¤¶È
int i=0;
int size = 0;
for(; i<readLen-MAX_SYNCCODE_LEN; i++)
{
if(!findSyncCode_0xFFF(&pFrame[i], &size))
{
continue;
}
else
{
break;
}
}
if(i!=0) // ²»ÊÇÖ¡¿ªÍ·£¬Æ«ÒƵ½Ö¡¿ªÍ·ÖØжÁ
{
printf("[%s %d]synccode error, i=%d\n",__FILE__,__LINE__,i);
free(pFrame);
rewind = (-(readLen-i));
fseek (pAACInfo->pFileFd, rewind, SEEK_CUR);
return -1;
}
// 3.¶ÁÈ¡ADTSÖ¡Êý¾Ý 2025-05-22 21:44:39
readLen = fread(pFrame+ADTS_HEADER_LEN, 1, size-ADTS_HEADER_LEN, pAACInfo->pFileFd);
if(readLen <= 0)
{
printf("[%s %d]fread error\n",__FILE__,__LINE__);
free(pFrame);
return -1;
}
// 4.ÌîÊý¾Ý
pAACFrame->frame_len = size;
pAACFrame->pFrameBuf = pFrame;
return pAACFrame->frame_len;
}
3、tcp_rtp.h
#ifndef _RTP_H_
#define _RTP_H_
#include <stdint.h>
#define RTP_VESION 2
#define RTP_PAYLOAD_TYPE_H264 96
#define RTP_PAYLOAD_TYPE_AAC 97
#define RTP_HEADER_SIZE 12
#define RTP_MAX_PKT_SIZE 1400
/*
*
* 0 1 2 3
* 7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |V=2|P|X| CC |M| PT | sequence number |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | timestamp |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | synchronization source (SSRC) identifier |
* +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
* | contributing source (CSRC) identifiers |
* : .... :
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
*/
struct RtpHeader
{
/* byte 0 */
uint8_t csrcLen:4;
uint8_t extension:1;
uint8_t padding:1;
uint8_t version:2;
/* byte 1 */
uint8_t payloadType:7;
uint8_t marker:1;
/* bytes 2,3 */
uint16_t seq;
/* bytes 4-7 */
uint32_t timestamp;
/* bytes 8-11 */
uint32_t ssrc;
};
struct RtpPacket
{
char header[4];
struct RtpHeader rtpHeader;
uint8_t payload[0];
};
void rtpHeaderInit(struct RtpPacket* rtpPacket, uint8_t csrcLen, uint8_t extension,
uint8_t padding, uint8_t version, uint8_t payloadType, uint8_t marker,
uint16_t seq, uint32_t timestamp, uint32_t ssrc);
int rtpSendPacket(int socket, uint8_t rtpChannel, struct RtpPacket* rtpPacket, uint32_t dataSize);
#endif //_RTP_H_
4、tcp_rtp.c
#include <sys/types.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include "tcp_rtp.h"
void rtpHeaderInit(struct RtpPacket* rtpPacket, uint8_t csrcLen, uint8_t extension,
uint8_t padding, uint8_t version, uint8_t payloadType, uint8_t marker,
uint16_t seq, uint32_t timestamp, uint32_t ssrc)
{
rtpPacket->rtpHeader.csrcLen = csrcLen;
rtpPacket->rtpHeader.extension = extension;
rtpPacket->rtpHeader.padding = padding;
rtpPacket->rtpHeader.version = version;
rtpPacket->rtpHeader.payloadType = payloadType;
rtpPacket->rtpHeader.marker = marker;
rtpPacket->rtpHeader.seq = seq;
rtpPacket->rtpHeader.timestamp = timestamp;
rtpPacket->rtpHeader.ssrc = ssrc;
}
int rtpSendPacket(int socket, uint8_t rtpChannel, struct RtpPacket* rtpPacket, uint32_t dataSize)
{
int ret;
rtpPacket->header[0] = '$';
rtpPacket->header[1] = rtpChannel;
rtpPacket->header[2] = ((dataSize+RTP_HEADER_SIZE) & 0xFF00 ) >> 8;
rtpPacket->header[3] = (dataSize+RTP_HEADER_SIZE) & 0xFF;
rtpPacket->rtpHeader.seq = htons(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = htonl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = htonl(rtpPacket->rtpHeader.ssrc);
ret = send(socket, (void*)rtpPacket, dataSize+RTP_HEADER_SIZE+4, 0);
rtpPacket->rtpHeader.seq = ntohs(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = ntohl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = ntohl(rtpPacket->rtpHeader.ssrc);
return ret;
}
5、rtsp_aac_tcp_main.c
/**
* @file rtsp_aac_tcp_main.c
* @author : https://blog.csdn.net/wkd_007
* @brief
* @version 0.1
* @date 2025-07-16
*
* @copyright Copyright (c) 2025
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/stat.h>
#include <fcntl.h>
#include "tcp_rtp.h"
#include "aacReader.h"
#define AAC_FILE_NAME "test.aac"
#define RTSP_PORT 8554
#define MAX_CLIENTS 5
#define SESSION_ID 10086001
#define SESSION_TIMEOUT 60
typedef struct
{
int rtpSendFd;
int rtpPort;
int rtpChannel;
int bPlayFlag; // 播放标志
char *cliIp;
} RTP_Send_t;
typedef enum
{
RTP_NULL,
RTP_PLAY,
RTP_PLAYING,
RTP_STOP,
} RTP_PLAY_STATE;
static int createUdpSocket()
{
int fd = socket(AF_INET, SOCK_DGRAM, 0);
if (fd < 0)
return -1;
int on = 1;
setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, (const char *)&on, sizeof(on));
return fd;
}
static int rtpSendAACFrame(int socket, int rtpChannel,
struct RtpPacket *rtpPacket, uint8_t *frame, uint32_t frameSize)
{
int ret;
rtpPacket->payload[0] = 0x00;
rtpPacket->payload[1] = 0x10;
rtpPacket->payload[2] = (frameSize & 0x1FE0) >> 5; // 高8位
rtpPacket->payload[3] = (frameSize & 0x1F) << 3; // 低5位
memcpy(rtpPacket->payload + 4, frame, frameSize);
ret = rtpSendPacket(socket, rtpChannel, rtpPacket, frameSize + 4);
if (ret < 0)
{
printf("failed to send rtp packet\n");
return -1;
}
rtpPacket->rtpHeader.seq++;
return 0;
}
void *sendRtp(void *arg)
{
RTP_Send_t *pRtpSend = (RTP_Send_t *)arg;
int rtp_send_fd = pRtpSend->rtpSendFd;
int rtpChannel = pRtpSend->rtpChannel;
struct RtpPacket *rtpPacket = (struct RtpPacket *)malloc(sizeof(struct RtpPacket) + 1500);
rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_AAC, 1, 0, 0, 0x32411);
// aac
AACReaderInfo_t aacInfo;
if (AAC_FileOpen(AAC_FILE_NAME, &aacInfo) < 0)
{
printf("failed to open %s\n", AAC_FILE_NAME);
return NULL;
}
while (pRtpSend->bPlayFlag)
{
if (!AAC_IsEndOfFile(&aacInfo))
{
AACFrame_t aacFrame;
memset(&aacFrame, 0, sizeof(aacFrame));
AAC_GetADTSFrame(&aacFrame, &aacInfo);
if (aacFrame.pFrameBuf != NULL)
{
// printf("rtpSendAACFrame\n");
rtpSendAACFrame(rtp_send_fd, rtpChannel, rtpPacket,
aacFrame.pFrameBuf + ADTS_HEADER_LEN, aacFrame.frame_len - ADTS_HEADER_LEN);
free(aacFrame.pFrameBuf);
/*
* 如果采样频率是48000
* 一般AAC每个1024个采样为一帧
* 所以一秒就有 48000 / 1024 = 46帧
* 时间增量就是 48000 / 46 = 1043
* 一帧的时间为 1000ms / 46 = 21ms
*/
rtpPacket->rtpHeader.timestamp += 1043;
usleep(21 * 1000);
}
else
{
printf("warning SeekFile\n");
AAC_SeekFile(&aacInfo);
}
}
}
free(rtpPacket);
AAC_FileClose(&aacInfo);
return NULL;
}
// 解析RTSP请求
void rtsp_request_parse(char *buffer, char *method, char *url, int *cseq, int *pRtpChannel)
{
char *line = strtok(buffer, "\r\n");
sscanf(line, "%s %s RTSP/1.0", method, url);
while ((line = strtok(NULL, "\r\n")) != NULL)
{
if (strncmp(line, "CSeq:", 5) == 0)
{
sscanf(line, "CSeq: %d", cseq);
}
char *pInterleaved = strstr(line, "interleaved=");
if (pInterleaved != NULL)
{
int rtcpChn = 0;
sscanf(pInterleaved, "interleaved=%d-%d", pRtpChannel, &rtcpChn);
// printf("rtpPort: %d-%d\n",*pRtpChannel, rtcpChn);
}
}
}
// 生成SDP描述
const char *generate_sdp()
{
return "v=0\r\n"
"o=- 0 0 IN IP4 0.0.0.0\r\n"
"s=Example Stream\r\n"
"t=0 0\r\n"
"m=audio 0 RTP/AVP 97\r\n"
"a=rtpmap:97 mpeg4-generic/48000/2\r\n"
"a=fmtp:97 SizeLength=13;\r\n"
"a=control:streamid=0\r\n";
}
void rtsp_handle_OPTION(char *response, int cseq)
{
sprintf(response,
"RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Public: OPTIONS, DESCRIBE, SETUP, PLAY, TEARDOWN\r\n\r\n",
cseq);
}
static void rtsp_handle_DESCRIBE(char *response, int cseq)
{
sprintf(response,
"RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: %zu\r\n\r\n%s",
cseq, strlen(generate_sdp()), generate_sdp());
}
static void rtsp_handle_SETUP(char *response, int cseq, int rtpChannel)
{
sprintf(response,
"RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Session: %u; timeout=%d\r\n"
"Transport: RTP/AVP/TCP;unicast;interleaved=%hhu-%hhu\r\n\r\n",
cseq, SESSION_ID, SESSION_TIMEOUT, rtpChannel, rtpChannel + 1);
}
static void rtsp_handle_PLAY(char *response, int cseq)
{
sprintf(response,
"RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Session: %u; timeout=%d\r\n"
"Range: npt=0.000-\r\n\r\n",
cseq, SESSION_ID, SESSION_TIMEOUT);
}
static void rtsp_handle_TEARDOWN(char *response, int cseq)
{
sprintf(response,
"RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Session: %d; timeout=%d\r\n\r\n",
cseq, SESSION_ID, SESSION_TIMEOUT);
}
// 处理客户端连接
int handle_client(int cli_fd, char *cli_ip)
{
int client_sock = cli_fd;
char buffer[1024] = {0};
int cseq = 0;
int rtpChn = 0;
unsigned char bSendFlag = RTP_NULL;
RTP_Send_t rtpSend;
pthread_t thread_id;
while (1)
{
memset(buffer, 0, sizeof(buffer));
int len = read(client_sock, buffer, sizeof(buffer) - 1);
if (len <= 0)
break;
printf("C->S [%s]\n\n", buffer);
char method[16] = {0};
char url[128] = {0};
rtsp_request_parse(buffer, method, url, &cseq, &rtpChn);
char response[1024] = {0}; // 构造响应
if (strcmp(method, "OPTIONS") == 0)
{
rtsp_handle_OPTION(response, cseq);
}
else if (strcmp(method, "DESCRIBE") == 0)
{
rtsp_handle_DESCRIBE(response, cseq);
}
else if (strcmp(method, "SETUP") == 0)
{
rtsp_handle_SETUP(response, cseq, rtpChn);
}
else if (strcmp(method, "PLAY") == 0)
{
rtsp_handle_PLAY(response, cseq);
bSendFlag = RTP_PLAY;
}
else if (strcmp(method, "TEARDOWN") == 0)
{
rtsp_handle_TEARDOWN(response, cseq);
bSendFlag = RTP_STOP;
}
else
{
snprintf(response, sizeof(response),
"RTSP/1.0 501 Not Implemented\r\nCSeq: %d\r\n\r\n", cseq);
}
write(client_sock, response, strlen(response));
printf("S->C [%s]\n\n", response);
if (bSendFlag == RTP_PLAY) // PLAY
{
rtpSend.rtpSendFd = cli_fd;
rtpSend.rtpPort = 0;
rtpSend.rtpChannel = rtpChn;
rtpSend.cliIp = NULL;
rtpSend.bPlayFlag = 1;
// 这里不使用线程的话，会一直无法处理 client_sock 发过来的 OPTION 消息，导致播放出问题
if (pthread_create(&thread_id, NULL, (void *)sendRtp, (void *)&rtpSend) < 0)
{
perror("pthread_create");
}
bSendFlag = RTP_PLAYING;
}
if (bSendFlag == RTP_STOP) // TEARDOWN
{
rtpSend.bPlayFlag = 0;
pthread_join(thread_id); // 等待线程结束
bSendFlag = RTP_NULL;
break;
}
}
printf("close ip=[%s] fd=[%d]\n", cli_ip, client_sock);
close(client_sock);
return 0;
}
int main(int argc, char *argv[])
{
int server_fd, client_fd;
struct sockaddr_in address;
int opt = 1;
socklen_t addrlen = sizeof(address);
// 创建套接字
if ((server_fd = socket(AF_INET, SOCK_STREAM, 0)) == 0)
{
perror("socket failed");
return -1;
}
// 设置套接字选项
if (setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &opt, sizeof(opt)))
{
perror("setsockopt");
return -1;
}
address.sin_family = AF_INET;
address.sin_addr.s_addr = INADDR_ANY;
address.sin_port = htons(RTSP_PORT);
// 绑定端口
if (bind(server_fd, (struct sockaddr *)&address, sizeof(address)) < 0)
{
perror("bind failed");
return -1;
}
// 开始监听
if (listen(server_fd, MAX_CLIENTS) < 0)
{
perror("listen");
return -1;
}
printf("RTSP Server listening on port %d\n", RTSP_PORT);
// 主循环接受连接，目前处理一个客户端
while (1)
{
char cli_ip[40] = {0};
if ((client_fd = accept(server_fd, (struct sockaddr *)&address, &addrlen)) < 0)
{
perror("accept");
return -1;
}
strncpy(cli_ip, inet_ntoa(address.sin_addr), sizeof(cli_ip));
printf("handle cliend [%s]\n", cli_ip);
handle_client(client_fd, cli_ip);
}
return 0;
}
首先设置一下VLC，工具->偏好设置->输入/编解码器，勾选如下图的RTP over RTSP（TCP）
将上面代码保存在同一个目录后，并且在同目录里放一个.aac文件，然后运行 gcc *.c -lpthread 编译，再执行./a.out运行程序，下面是我运行的过程：
🎄四、总结
👉本文介绍了RTP_over_TCP的一些概念，以及TCP传输AAC格式的RTP包的RTSP服务器实现的步骤和细节，最后提供了实现的源代码，帮助读者学习理解。
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