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从零开始写一个RTSP服务器（四）一个传输H.264的RTSP服务器_rtspserver
作者：羊村懒王 | 2024-05-23 03:30:40
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rtspserver
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（四）一个传输H.264的RTSP服务器
本篇文章的目的：这篇文章的目的是实现一个播放H.264的RTSP服务器，相信如果是仔细看了前面几篇文章的话，这简直是易如反掌，就是把前面的东西拼到一起
所以这篇文章并不会讲述新的知识，只是把前面的东西拼凑到一起，整理一下思路，最后给出一个示例，下面开始讲解一下我提供的这个示例的运行流程
一、建立套接字
一开始进入main函数后，就监听服务器tcp套接字，绑定端口号，然后开始监听
然后再分别建立用于RTP和RTCP的udp套接字，绑定好端口
然后进入循环中开始服务
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202 */ main() { /* 创建服务器tcp套接字，绑定端口，监听 */ serverSockfd = createTcpSocket(); bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT); listen(serverSockfd, 10); /* 建立用于RTP和RTCP的udp套接字，绑定好端口 */ serverRtpSockfd = createUdpSocket(); serverRtcpSockfd = createUdpSocket(); bindSocketAddr(serverRtpSockfd, "0.0.0.0", SERVER_RTP_PORT); bindSocketAddr(serverRtcpSockfd, "0.0.0.0", SERVER_RTCP_PORT); while(1) { ... } }
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二、接收客户端连接
在while循环中接收客户端，然后调用doClient服务
main()
{
...
while(1)
{
clientSockfd = acceptClient(serverSockfd, clientIp, &clientPort);
doClient(clientSockfd, clientIp, clientPort, serverRtpSockfd, serverRtcpSockfd);
}
}
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上面其实就是一个TCP服务器的基本步骤，没有什么特别的
下面来看一看doClient函数
三、解析命令
doClient就是一个while循环（这是一个同时只能服务一个客户的服务器），不断地接收命令解析命令，然后调用相应地操作
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202 */ doClient() { while(1) { recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0); ... sscanf(line, "%s %s %s\r\n", method, url, version); ... sscanf(line, "CSeq: %d\r\n", &cseq) ... } }
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四、处理请求
在解析完客户端命令后，会调用相应的请求，处理完之后讲接收打印到sBuf中，然后发送给客户端
doClient() { while(1) { ... /* 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, clientRtpPort); else if(!strcmp(method, "PLAY")) handleCmd_PLAY(sBuf, cseq); /* 放回结果 */ send(clientSockfd, sBuf, strlen(sBuf), 0); } }
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下面来看看各个请求的行动
4.1 OPTIONS
返回可用方法
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
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;
}
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4.2 DESCRIBE
返回sdp文件信息，这是一个H.264的媒体描述信息，详细内容已经在前面文章讲解过，这里不再累赘
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202 */ 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" "m=video 0 RTP/AVP 96\r\n" "a=rtpmap:96 H264/90000\r\n" "a=control:track0\r\n", time(NULL), localIp); 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; }
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4.3 SETUP
SETUP过程发送服务端RTP端口和RTCP端口
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202 */ static int handleCmd_SETUP(char* result, int cseq, int clientRtpPort, int* localRtpSockfd, int* localRtcpSockfd) { sprintf(result, "RTSP/1.0 200 OK\r\n" "CSeq: %d\r\n" "Transport: RTP/AVP;unicast;client_port=%d-%d;server_port=%d-%d\r\n" "Session: 66334873\r\n" "\r\n", cseq, clientRtpPort, clientRtpPort+1, SERVER_RTP_PORT, SERVER_RTCP_PORT); return 0; }
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4.4 PLAY
PLAY操作回复后，会开始发送RTP包
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
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;
}
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五、H.264 RTP打包发送
从H.264文件中读取一个NALU，向客户端发送RTP包（目的IP，目的RTP端口）
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202 */ doClient() { while(1) { ... ... ... if(!strcmp(method, "PLAY")) { while(1) { /* 获取一帧 */ frameSize = getFrameFromH264File(fd, frame, 500000); /* RTP打包发送 */ rtpSendH264Frame(localRtpSockfd, clientIP, clientRtpPort, rtpPacket, frame+startCode, frameSize); } } } }
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下面看一看RTP打包过程，RTP打包实现了单NALU打包和分片打包
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202 */ static int rtpSendH264Frame(int socket, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize) { /* 如果包比较小，则采用单NALU打包 */ if (frameSize <= RTP_MAX_PKT_SIZE) { rtpSendPacket(socket, ip, port, rtpPacket, frameSize); } else //否则采用分片打包 { for (i = 0; i < pktNum; i++) { /* 填充载荷的前两个字节 */ rtpPacket->payload[0] = (naluType & 0x60) | 28; rtpPacket->payload[1] = naluType & 0x1F; /* 发送RTP包 */ rtpSendPacket(socket, ip, port, rtpPacket, RTP_MAX_PKT_SIZE+2); } } }
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六、源码
我也懒得建个Git仓了，源码就直接贴到这里吧，虽然有点长，嘻嘻
总共有3个文件，h264_rtsp_server.c、rtp.c、rtp.h
h264_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 "rtp.h" #define H264_FILE_NAME "test.h264" #define SERVER_PORT 8554 #define SERVER_RTP_PORT 55532 #define SERVER_RTCP_PORT 55533 #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; return -1; } 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" "m=video 0 RTP/AVP 96\r\n" "a=rtpmap:96 H264/90000\r\n" "a=control:track0\r\n", time(NULL), localIp); 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, int clientRtpPort) { sprintf(result, "RTSP/1.0 200 OK\r\n" "CSeq: %d\r\n" "Transport: RTP/AVP;unicast;client_port=%d-%d;server_port=%d-%d\r\n" "Session: 66334873\r\n" "\r\n", cseq, clientRtpPort, clientRtpPort+1, SERVER_RTP_PORT, SERVER_RTCP_PORT); 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, int serverRtpSockfd, int serverRtcpSockfd) { char method[40]; char url[100]; char version[40]; int cseq; int clientRtpPort, clientRtcpPort; 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; } /* 如果是SETUP，那么就再解析client_port */ if(!strcmp(method, "SETUP")) { while(1) { bufPtr = getLineFromBuf(bufPtr, line); if(!strncmp(line, "Transport:", strlen("Transport:"))) { sscanf(line, "Transport: RTP/AVP;unicast;client_port=%d-%d\r\n", &clientRtpPort, &clientRtcpPort); break; } } } 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")) { if(handleCmd_SETUP(sBuf, cseq, clientRtpPort)) { 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); /* 开始播放，发送RTP包 */ if(!strcmp(method, "PLAY")) { int frameSize, startCode; char* frame = malloc(500000); struct RtpPacket* rtpPacket = (struct RtpPacket*)malloc(500000); int 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); printf("start play\n"); printf("client ip:%s\n", clientIP); printf("client port:%d\n", clientRtpPort); while (1) { frameSize = getFrameFromH264File(fd, frame, 500000); if(frameSize < 0) { break; } if(startCode3(frame)) startCode = 3; else startCode = 4; frameSize -= startCode; rtpSendH264Frame(serverRtpSockfd, clientIP, clientRtpPort, rtpPacket, frame+startCode, frameSize); rtpPacket->rtpHeader.timestamp += 90000/25; usleep(1000*1000/25); } free(frame); free(rtpPacket); goto out; } } out: printf("finish\n"); close(clientSockfd); free(rBuf); free(sBuf); } int main(int argc, char* argv[]) { int serverSockfd; int serverRtpSockfd, serverRtcpSockfd; serverSockfd = createTcpSocket(); if(serverSockfd < 0) { printf("failed to create tcp socket\n"); return -1; } if(bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT) < 0) { printf("failed to bind addr\n"); return -1; } if(listen(serverSockfd, 10) < 0) { printf("failed to listen\n"); return -1; } serverRtpSockfd = createUdpSocket(); serverRtcpSockfd = createUdpSocket(); if(serverRtpSockfd < 0 || serverRtcpSockfd < 0) { printf("failed to create udp socket\n"); return -1; } if(bindSocketAddr(serverRtpSockfd, "0.0.0.0", SERVER_RTP_PORT) < 0 || bindSocketAddr(serverRtcpSockfd, "0.0.0.0", SERVER_RTCP_PORT) < 0) { printf("failed to bind addr\n"); return -1; } printf("rtsp://127.0.0.1:%d\n", SERVER_PORT); 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, serverRtpSockfd, serverRtcpSockfd); } return 0; }
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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; }
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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_
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七、测试
将三个文件保存下来，h264_rtsp_server.c、rtp.c、rtp.h
编译
# gcc h264_rtsp_server.c rtp.c
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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
# ./a.out
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运行后会打印一个url
rtsp://127.0.0.1:8554
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在vlc中输入url，即可看到视频
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运行效果
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