7、多播的RTSP服务器 |
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7、多播的RTSP服务器
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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 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\nS—>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 232.123.86.248/255 c=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
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 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 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 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #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 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); 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 #ifndef _RTP_H_ #define _RTP_H_ #include #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 #include #include #include #include #include #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
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