从零开始写一个RTSP服务器系列
从零开始写一个RTSP服务器（四）一个传输H.264的RTSP服务器
从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器
从零开始写一个RTSP服务器（九）一个RTP OVER RTSP/TCP的RTSP服务器
从零开始写一个RTSP服务器（八）一个多播的RTSP服务器
文章目录
本文目的：实现一个多播传输H.264的RTSP服务器
一、多播的RTSP交互过程
我们在前面文章从零开始写一个RTSP服务器（二）RTSP协议的实现中实现的RTSP交互过程是RTSP单播的情况，多播的实现过程与单播还是有所区别的，多播并不需要为每一个RTP和RTCP建立新的UDP套接字，只需要持续向一个多播地址推送RTP包就行，下面来看一个RTSP多播交互的示例
OPTIONS
-
C–>S
OPTIONS rtsp://127.0.0.1:8554/live RTSP/1.0\r\n CSeq: 2\r\n \r\n -
S–>C
RTSP/1.0 200 OK\r\n CSeq: 2\r\n Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY\r\n \r\n
DESCRIBE
-
C–>S
DESCRIBE rtsp://127.0.0.1:8554/live RTSP/1.0\r\n CSeq: 3\r\n Accept: application/sdp\r\n \r\n -
S—>C
RTSP/1.0 200 OK\r\n CSeq: 3\r\n Content-length: 225\r\n Content-type: application/sdp\r\n \r\n v=0 o=- 91565615172 1 IN IP4 127.0.0.1 t=0 0 a=control:* a=type:broadcast a=rtcp-unicast: reflection m=video 39016 RTP/AVP 96 c=IN IP4 232.123.86.248/255 a=rtpmap:96 H264/90000 a=framerate:25 a=control:track0这个sdp描述文件里指明了多播地址和多播端口
-
这一行表明RTCP反馈采用单播
a=rtcp-unicast: reflection在多播的情况下，这表明服务端RTP发送到多播组，RTCP发送到多播组，RTCP接收采用单播
-
这一行表明的多播目的端口为39016
m=video 39016 RTP/AVP 96 -
这一行表明了多播地址为
c=IN IP4 232.123.86.248/255c=IN IP4 232.123.86.248/255
-
多播和单播的sdp文件区别主要是多播需要指定好多播地址和多播端口
关于sdp这里就不再详细讲述了，如何有不清楚请看前面的文章
SETUP
-
C–>S
SETUP rtsp://127.0.0.1:8554/live/track0 RTSP/1.0\r\n CSeq: 4\r\n Transport: RTP/AVP;multicast;client_port=39016-39017 \r\n -
S–>C
RTSP/1.0 200 OK\r\n CSeq: 4\r\n Transport: RTP/AVP;multicast;destination=232.123.86.248;source=192.168.31.115;port=39016-39017;ttl=255 Session: 66334873 \r\n
PLAY
-
C–>S
PLAY rtsp://127.0.0.1:8554/live RTSP/1.0\r\n CSeq: 5\r\n Session: 66334873 Range: npt=0.000-\r\n \r\n -
S–>C
RTSP/1.0 200 OK\r\n CSeq: 5\r\n Range: npt=0.000-\r\n Session: 66334873; timeout=60 \r\n
二、多播的RTSP服务器实现过程
2.1 创建套接字
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
main()
{
/* 创建套接字 */
serverSockfd = createTcpSocket();
/* 绑定地址和端口 */
bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
/* 开始监听 */
listen(serverSockfd, 10);
...
while(1)
{
...
}
}
2.2 创建线程向多播地址推送RTP包
在进入while循环接收客户端前，我们创建一个线程不断地向多播地址发送RTP包
main()
{
...
pthread_create(&threadId, NULL, sendRtpPacket, NULL);
while(1)
{
...
}
}
下面看一看发送函数
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
sendRtpPacket()
{
...
while(1)
{
...
/* 获取一帧数据 */
getFrameFromH264File(fd, frame, 500000);
/* 向多播地址发送RTP包 */
rtpSendH264Frame(sockfd, MULTICAST_IP, MULTICAST_PORT,
rtpPacket, frame+startCode, frameSize);
...
}
}
2.2 接收客户端连接
进入while循环后，开始接收客户端，然后处理客户端请求
main()
{
...
while(1)
{
/* 接收客户端 */
acceptClient(serverSockfd, clientIp, &clientPort);
/* 处理客户端 */
doClient(clientSockfd, clientIp, clientPort);
}
}
下面仔细看一看如何处理客户端请求
2.3 解析命令
先解析请求方法，然后解析序列号，再根据不同地请求做不同的处理，然后再放回结果给客户端
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
doClient()
{
...
while(1)
{
/* 接收请求 */
recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
...
/* 解析请求方法 */
sscanf(line, "%s %s %s\r\n", method, url, version)
...
/* 解析序列号 */
sscanf(line, "CSeq: %d\r\n", &cseq);
/* 处理请求 */
if(!strcmp(method, "OPTIONS"))
handleCmd_OPTIONS(sBuf, cseq);
else if(!strcmp(method, "DESCRIBE"))
handleCmd_DESCRIBE(sBuf, cseq, url);
else if(!strcmp(method, "SETUP"))
handleCmd_SETUP(sBuf, cseq, localIp);
else if(!strcmp(method, "PLAY"))
handleCmd_PLAY(sBuf, cseq);
/* 返回结果给客户端 */
send(clientSockfd, sBuf, strlen(sBuf), 0);
}
}
2.4 处理请求
-
OPTIONS
handleCmd_OPTIONS() { sprintf(result, "RTSP/1.0 200 OK\r\n" "CSeq: %d\r\n" "Public: OPTIONS, DESCRIBE, SETUP, PLAY\r\n" "\r\n", cseq); } -
DESCRIBE
发送多播的sdp描述文件
handleCmd_DESCRIBE() { /* 多播sdp */ sprintf(sdp, "v=0\r\n" "o=- 9%ld 1 IN IP4 %s\r\n" "t=0 0\r\n" "a=control:*\r\n" "a=type:broadcast\r\n" "a=rtcp-unicast: reflection\r\n" "m=video %d RTP/AVP 96\r\n" "c=IN IP4 %s/255\r\n" "a=rtpmap:96 H264/90000\r\n" "a=control:track0\r\n", time(NULL), localIp, MULTICAST_PORT, MULTICAST_IP); sprintf(result, "RTSP/1.0 200 OK\r\nCSeq: %d\r\n" "Content-Base: %s\r\n" "Content-type: application/sdp\r\n" "Content-length: %d\r\n\r\n" "%s", cseq, url, strlen(sdp), sdp); } -
SETUP
handleCmd_SETUP() { sprintf(result, "RTSP/1.0 200 OK\r\n" "CSeq: %d\r\n" "Transport: RTP/AVP;multicast;destination=%s;" "source=%s;port=%d- %d;ttl=255\r\n" "Session: 66334873\r\n" "\r\n", cseq, MULTICAST_IP, localIp, MULTICAST_PORT, MULTICAST_PORT+1); } -
PLAY
handleCmd_PLAY() { sprintf(result, "RTSP/1.0 200 OK\r\n" "CSeq: %d\r\n" "Range: npt=0.000-\r\n" "Session: 66334873; timeout=60\r\n\r\n", cseq); }在play回复完成之后，客户端就会去多播地址拉取媒体流，然后播放
源码
源码有三个文件：multicast_rtsp_server、rtp.h、rtp.c
multicast_rtsp_server.c
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <assert.h>
#include <pthread.h>
#include "rtp.h"
#define H264_FILE_NAME "test.h264"
#define MULTICAST_IP "239.255.255.11"
#define SERVER_PORT 8554
#define MULTICAST_PORT 9832
#define BUF_MAX_SIZE (1024*1024)
static int createTcpSocket()
{
int sockfd;
int on = 1;
sockfd = socket(AF_INET, SOCK_STREAM, 0);
if(sockfd < 0)
return -1;
setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on));
return sockfd;
}
static int createUdpSocket()
{
int sockfd;
int on = 1;
sockfd = socket(AF_INET, SOCK_DGRAM, 0);
if(sockfd < 0)
return -1;
setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on));
return sockfd;
}
static int bindSocketAddr(int sockfd, const char* ip, int port)
{
struct sockaddr_in addr;
addr.sin_family = AF_INET;
addr.sin_port = htons(port);
addr.sin_addr.s_addr = inet_addr(ip);
if(bind(sockfd, (struct sockaddr *)&addr, sizeof(struct sockaddr)) < 0)
return -1;
return 0;
}
static int acceptClient(int sockfd, char* ip, int* port)
{
int clientfd;
socklen_t len = 0;
struct sockaddr_in addr;
memset(&addr, 0, sizeof(addr));
len = sizeof(addr);
clientfd = accept(sockfd, (struct sockaddr *)&addr, &len);
if(clientfd < 0)
return -1;
strcpy(ip, inet_ntoa(addr.sin_addr));
*port = ntohs(addr.sin_port);
return clientfd;
}
static inline int startCode3(char* buf)
{
if(buf[0] == 0 && buf[1] == 0 && buf[2] == 1)
return 1;
else
return 0;
}
static inline int startCode4(char* buf)
{
if(buf[0] == 0 && buf[1] == 0 && buf[2] == 0 && buf[3] == 1)
return 1;
else
return 0;
}
static char* findNextStartCode(char* buf, int len)
{
int i;
if(len < 3)
return NULL;
for(i = 0; i < len-3; ++i)
{
if(startCode3(buf) || startCode4(buf))
return buf;
++buf;
}
if(startCode3(buf))
return buf;
return NULL;
}
static int getFrameFromH264File(int fd, char* frame, int size)
{
int rSize, frameSize;
char* nextStartCode;
if(fd < 0)
return fd;
rSize = read(fd, frame, size);
if(!startCode3(frame) && !startCode4(frame))
return -1;
nextStartCode = findNextStartCode(frame+3, rSize-3);
if(!nextStartCode)
{
lseek(fd, 0, SEEK_SET);
frameSize = rSize;
}
else
{
frameSize = (nextStartCode-frame);
lseek(fd, frameSize-rSize, SEEK_CUR);
}
return frameSize;
}
static int rtpSendH264Frame(int socket, const char* ip, int16_t port,
struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize)
{
uint8_t naluType; // nalu第一个字节
int sendBytes = 0;
int ret;
naluType = frame[0];
if (frameSize <= RTP_MAX_PKT_SIZE) // nalu长度小于最大包场：单一NALU单元模式
{
/*
* 0 1 2 3 4 5 6 7 8 9
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |F|NRI| Type | a single NAL unit ... |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
memcpy(rtpPacket->payload, frame, frameSize);
ret = rtpSendPacket(socket, ip, port, rtpPacket, frameSize);
if(ret < 0)
return -1;
rtpPacket->rtpHeader.seq++;
sendBytes += ret;
if ((naluType & 0x1F) == 7 || (naluType & 0x1F) == 8) // 如果是SPS、PPS就不需要加时间戳
goto out;
}
else // nalu长度小于最大包场：分片模式
{
/*
* 0 1 2
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | FU indicator | FU header | FU payload ... |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/*
* FU Indicator
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F|NRI| Type |
* +---------------+
*/
/*
* FU Header
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |S|E|R| Type |
* +---------------+
*/
int pktNum = frameSize / RTP_MAX_PKT_SIZE; // 有几个完整的包
int remainPktSize = frameSize % RTP_MAX_PKT_SIZE; // 剩余不完整包的大小
int i, pos = 1;
/* 发送完整的包 */
for (i = 0; i < pktNum; i++)
{
rtpPacket->payload[0] = (naluType & 0x60) | 28;
rtpPacket->payload[1] = naluType & 0x1F;
if (i == 0) //第一包数据
rtpPacket->payload[1] |= 0x80; // start
else if (remainPktSize == 0 && i == pktNum - 1) //最后一包数据
rtpPacket->payload[1] |= 0x40; // end
memcpy(rtpPacket->payload+2, frame+pos, RTP_MAX_PKT_SIZE);
ret = rtpSendPacket(socket, ip, port, rtpPacket, RTP_MAX_PKT_SIZE+2);
if(ret < 0)
return -1;
rtpPacket->rtpHeader.seq++;
sendBytes += ret;
pos += RTP_MAX_PKT_SIZE;
}
/* 发送剩余的数据 */
if (remainPktSize > 0)
{
rtpPacket->payload[0] = (naluType & 0x60) | 28;
rtpPacket->payload[1] = naluType & 0x1F;
rtpPacket->payload[1] |= 0x40; //end
memcpy(rtpPacket->payload+2, frame+pos, remainPktSize+2);
ret = rtpSendPacket(socket, ip, port, rtpPacket, remainPktSize+2);
if(ret < 0)
return -1;
rtpPacket->rtpHeader.seq++;
sendBytes += ret;
}
}
out:
return sendBytes;
}
static char* getLineFromBuf(char* buf, char* line)
{
while(*buf != '\n')
{
*line = *buf;
line++;
buf++;
}
*line = '\n';
++line;
*line = '\0';
++buf;
return buf;
}
static int handleCmd_OPTIONS(char* result, int cseq)
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Public: OPTIONS, DESCRIBE, SETUP, PLAY\r\n"
"\r\n",
cseq);
return 0;
}
static int handleCmd_DESCRIBE(char* result, int cseq, char* url)
{
char sdp[500];
char localIp[100];
sscanf(url, "rtsp://%[^:]:", localIp);
sprintf(sdp, "v=0\r\n"
"o=- 9%ld 1 IN IP4 %s\r\n"
"t=0 0\r\n"
"a=control:*\r\n"
"a=type:broadcast\r\n"
"a=rtcp-unicast: reflection\r\n"
"m=video %d RTP/AVP 96\r\n"
"c=IN IP4 %s/255\r\n"
"a=rtpmap:96 H264/90000\r\n"
"a=control:track0\r\n",
time(NULL),
localIp,
MULTICAST_PORT,
MULTICAST_IP);
sprintf(result, "RTSP/1.0 200 OK\r\nCSeq: %d\r\n"
"Content-Base: %s\r\n"
"Content-type: application/sdp\r\n"
"Content-length: %d\r\n\r\n"
"%s",
cseq,
url,
strlen(sdp),
sdp);
return 0;
}
static int handleCmd_SETUP(char* result, int cseq, char* localIp)
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Transport: RTP/AVP;multicast;destination=%s;source=%s;port=%d-%d;ttl=255\r\n"
"Session: 66334873\r\n"
"\r\n",
cseq,
MULTICAST_IP,
localIp,
MULTICAST_PORT,
MULTICAST_PORT+1);
return 0;
}
static int handleCmd_PLAY(char* result, int cseq)
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Range: npt=0.000-\r\n"
"Session: 66334873; timeout=60\r\n\r\n",
cseq);
return 0;
}
static void doClient(int clientSockfd, const char* clientIP, int clientPort)
{
char method[40];
char url[100];
char localIp[40];
char version[40];
int cseq;
char *bufPtr;
char* rBuf = malloc(BUF_MAX_SIZE);
char* sBuf = malloc(BUF_MAX_SIZE);
char line[400];
while(1)
{
int recvLen;
recvLen = recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
if(recvLen <= 0)
goto out;
rBuf[recvLen] = '\0';
printf("---------------C->S--------------\n");
printf("%s", rBuf);
/* 解析方法 */
bufPtr = getLineFromBuf(rBuf, line);
if(sscanf(line, "%s %s %s\r\n", method, url, version) != 3)
{
printf("parse err\n");
goto out;
}
/* 解析序列号 */
bufPtr = getLineFromBuf(bufPtr, line);
if(sscanf(line, "CSeq: %d\r\n", &cseq) != 1)
{
printf("parse err\n");
goto out;
}
if(!strcmp(method, "OPTIONS"))
{
if(handleCmd_OPTIONS(sBuf, cseq))
{
printf("failed to handle options\n");
goto out;
}
}
else if(!strcmp(method, "DESCRIBE"))
{
if(handleCmd_DESCRIBE(sBuf, cseq, url))
{
printf("failed to handle describe\n");
goto out;
}
}
else if(!strcmp(method, "SETUP"))
{
sscanf(url, "rtsp://%[^:]:", localIp);
if(handleCmd_SETUP(sBuf, cseq, localIp))
{
printf("failed to handle setup\n");
goto out;
}
}
else if(!strcmp(method, "PLAY"))
{
if(handleCmd_PLAY(sBuf, cseq))
{
printf("failed to handle play\n");
goto out;
}
}
else
{
goto out;
}
printf("---------------S->C--------------\n");
printf("%s", sBuf);
send(clientSockfd, sBuf, strlen(sBuf), 0);
}
out:
printf("finish\n");
close(clientSockfd);
free(rBuf);
free(sBuf);
}
void* sendRtpPacket(void* arg)
{
int fd;
int frameSize, startCode;
char* frame = malloc(500000);
struct RtpPacket* rtpPacket = (struct RtpPacket*)malloc(500000);
int sockfd = createUdpSocket();
assert(sockfd > 0);
fd = open(H264_FILE_NAME, O_RDONLY);
assert(fd > 0);
rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_H264, 0,
0, 0, 0x88923423);
while(1)
{
frameSize = getFrameFromH264File(fd, frame, 500000);
if(startCode3(frame))
startCode = 3;
else
startCode = 4;
frameSize -= startCode;
rtpSendH264Frame(sockfd, MULTICAST_IP, MULTICAST_PORT,
rtpPacket, frame+startCode, frameSize);
rtpPacket->rtpHeader.timestamp += 90000/25;
usleep(1000*1000/25);
}
free(frame);
free(rtpPacket);
close(fd);
return NULL;
}
int main(int argc, char* argv[])
{
int serverSockfd;
int ret;
pthread_t threadId;
serverSockfd = createTcpSocket();
if(serverSockfd < 0)
{
printf("failed to create tcp socket\n");
return -1;
}
ret = bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
if(ret < 0)
{
printf("failed to bind addr\n");
return -1;
}
ret = listen(serverSockfd, 10);
if(ret < 0)
{
printf("failed to listen\n");
return -1;
}
printf("rtsp://127.0.0.1:%d\n", SERVER_PORT);
pthread_create(&threadId, NULL, sendRtpPacket, NULL);
while(1)
{
int clientSockfd;
char clientIp[40];
int clientPort;
clientSockfd = acceptClient(serverSockfd, clientIp, &clientPort);
if(clientSockfd < 0)
{
printf("failed to accept client\n");
return -1;
}
printf("accept client;client ip:%s,client port:%d\n", clientIp, clientPort);
doClient(clientSockfd, clientIp, clientPort);
}
return 0;
}
rtp.h
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
#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
{
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, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint32_t dataSize);
#endif //_RTP_H_
rtp.c
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
#include <sys/types.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include "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, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint32_t dataSize)
{
struct sockaddr_in addr;
int ret;
addr.sin_family = AF_INET;
addr.sin_port = htons(port);
addr.sin_addr.s_addr = inet_addr(ip);
rtpPacket->rtpHeader.seq = htons(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = htonl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = htonl(rtpPacket->rtpHeader.ssrc);
ret = sendto(socket, (void*)rtpPacket, dataSize+RTP_HEADER_SIZE, 0,
(struct sockaddr*)&addr, sizeof(addr));
rtpPacket->rtpHeader.seq = ntohs(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = ntohl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = ntohl(rtpPacket->rtpHeader.ssrc);
return ret;
}
三、测试
将multicast_rtsp_server.c、rtp.h、rtp.c保存下来
编译运行，程序默认会打开test.h264的视频文件，如果你没有视频源的话，可以从RtspServer的example目录下获取
# gcc multicast_rtsp_server.c rtp.c -lpthread
# ./a.out
运行后会打印一个url
rtsp://127.0.0.1:8554
在vlc中输入url，就可以播放了
-
运行效果
1040
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
