继续上篇介绍，本篇介绍一下封装RTP的数据格式，如何将摄像头采集的码流，音频的码流，封装到rtp里，传输。
有自己私有协议例子，有rtp协议，参考代码。注意不是rtsp协议。
一、私有协议
玩过tcp协议都知道，有这么一层关系
玩过网络开发的，应该都自己定义封装过私有协议。
我把tcp，udp，网络传输封装在另一个文件，网络传输层。然后分一个模块，私有协议，封装有效数据，数据封装层。
封装一个协议头
typedef struct MediumPrivateHead
{SX_U32 u32Mf;SX_U16 u16Type;SX_U32 u32Ts;SX_U16 u16Seqno;SX_U32 u32Length;SX_U8 u8Ch;SX_U8 u8Of;SX_U8 u8Marker;SX_U8 u8X;
}T_MediumPrivateHead, *PT_MediumPrivateHead;
#pragma pack()typedef enum PacketType
{PACKET_TYPE_H264 = 118,PACKET_TYPE_JPEG = 119,PACKET_TYPE_AAC = 120,PACKET_TYPE_G711 = 121,PACKET_TYPE_H265 = 122,PACKET_TYPE_ADPCM_DVI4 = 123,PACKET_TYPE_G721 = 124,PACKET_TYPE_RS422 = 125,PACKET_TYPE_BUTT = 126,
}E_PacketType;
分片发送，数据包太大，进行分包传输，不管是什么协议，都需要调用这个函数来分包传输，传入一个bufer数据指针。
static SX_U32 MEDIUM_UDP_WriteFrame1( SX_S32 ch, SX_S8 ePacketType, void *handle, SX_S8 *ps8Buffer, SX_U32 u32Length, SX_U64 u64Pts, SX_S8 *ps8TsString )
{PT_Medium ptTmp = (PT_Medium)handle;SX_U32 u32Pos = 0;SX_U32 u32LeftLen = u32Length;SX_S8 *ps8Tmp = ps8Buffer;int allow_send_max_len = 0;//if(ptTmp->u32FragmentLen < 300) // len too short//ptTmp->u32FragmentLen = 300;allow_send_max_len = ptTmp->u32FragmentLen - sizeof(T_MediumPrivateHead);if( u32LeftLen <= allow_send_max_len ){if( MEDIUM_UDP_WritePrivatePacket( ch, ePacketType, handle, ps8Tmp, u32LeftLen, u64Pts, 1, 0 ) < 0 )return 0; }else{while( u32LeftLen > allow_send_max_len ){if( MEDIUM_UDP_WritePrivatePacket( ch, ePacketType, handle, ps8Tmp + u32Pos, allow_send_max_len, u64Pts, 0, 0 ) < 0 )return 0;u32LeftLen -= allow_send_max_len;u32Pos += allow_send_max_len;}if( MEDIUM_UDP_WritePrivatePacket( ch, ePacketType, handle, ps8Tmp + u32Pos, u32LeftLen, u64Pts, 1, 0 ) < 0 )return 0;}return u32Length;
}封装私有协议包，供上面函数调用，如果是其他协议，就封装另一个函数。将有效数据和协议头封装，打包，通过udp或tcp的接口sendto发送出去
static SX_S32 MEDIUM_UDP_WritePrivatePacket( SX_S32 ch, SX_S8 ePacketType, void *handle, SX_S8 *ps8Buffer, SX_U32 u32Length, SX_U64 u64Pts, SX_U32 u32Marker, SX_U32 u32X )
{if(ch < 0 || ch >= MEDIUM_MAX_CH)return -1;SX_U32 u32Pos = 0;PT_Medium ptTmp = (PT_Medium)handle;pthread_mutex_lock( &ptTmp->mutex);SX_S8 *ps8Buf = ptTmp->ps8Buffer;memset(ps8Buf ,0 ,MAX_CONFIG_FILE_LEN);SX_U32 u32myframeLen;if( u32Length > ptTmp->u32FragmentLen - sizeof(T_MediumPrivateHead))u32myframeLen = ptTmp->u32FragmentLen- sizeof(T_MediumPrivateHead);elseu32myframeLen = u32Length;//file private head //PT_VedioDataPkt pkthead1 = (PT_VedioDataPkt)(ps8Buf);//PT_MediumPrivateHead ptHead2 = (PT_MediumPrivateHead)(&pkthead1->headprivate);PT_MediumPrivateHead ptHead2 = (PT_MediumPrivateHead)(ps8Buf);memset((SX_U8 *)ptHead2, 0, sizeof(T_MediumPrivateHead) );ptHead2->u32Mf = (SX_U32)0x4b4e4148;ptHead2->u16Type = (SX_U16)ePacketType;ptHead2->u32Ts = (SX_U32)u64Pts; //not usedptHead2->u16Seqno = ((ePacketType == PACKET_TYPE_AAC )? (SX_U16)ptTmp->u16AudioSeqno[ch] : (SX_U16)ptTmp- >u16VedioSeqno[ch]);ptHead2->u32Length = (SX_U32)u32myframeLen;ptHead2->u8Ch = (SX_U8)ch;ptHead2->u8Of = (SX_U8)0x19;ptHead2->u8Marker = (SX_U8)u32Marker;ptHead2->u8X = (SX_U8)u32X;u32Pos = sizeof(T_MediumPrivateHead);//filevedio frame datamemcpy( ps8Buf + u32Pos, ps8Buffer, u32myframeLen );//CRC32 = 0xC4C3C2C1;*(ps8Buf + u32Pos + u32myframeLen + 0)= 0xC1;*(ps8Buf + u32Pos + u32myframeLen + 1)= 0xC2;*(ps8Buf + u32Pos + u32myframeLen + 2)= 0xC3;*(ps8Buf + u32Pos + u32myframeLen + 3)= 0xC4;if(ePacketType == PACKET_TYPE_AAC){//printf("++ u16AudioSeqno %ld u32Length %ld ++\r\n",ptTmp->u16AudioSeqno[ch],sizeof(T_MediumPrivateHead) + ptHead2->u32Length + 4);ptTmp->u16AudioSeqno[ch]++;}else{ptTmp->u16VedioSeqno[ch]++;}if(ptTmp->u16AudioSeqno[ch] > 65536)ptTmp->u16AudioSeqno[ch] = 0;if(ptTmp->u16VedioSeqno[ch] > 65536)ptTmp->u16VedioSeqno[ch] = 0;
#endifpthread_mutex_unlock( &ptTmp->mutex);ts1 = get_sys_ms();if( sendto(ptTmp->fd,ps8Buf,sizeof(T_MediumPrivateHead) + u32myframeLen + 4, //headlen + datalen + crclen,0,(struct sockaddr *)&ptTmp->other[(ePacketType == PACKET_TYPE_AAC) ? 1 : 0],sizeof(struct sockaddr_in)) < 0 ) TRACE( DL_WARNING, "udp send failed\n" );return 0;
}二、 RTP协议
RTP传输音视频过程如下：
如果不按我上面私有协议传输，那就需要封装一个RTP协议，要熟悉协议格式，进行封装。
RTP报文格式
用C语言来封装，如下
取一段抓包数据
三、RTP封装视频
3.1、RTP封装H264
首先看一下H264 NALU头部定义：一个字节，是8位，按位进行划分，代表的意义。
F: 1 个比特. 一般为0 forbidden_zero_bit. 在 H.264 规范中规定了这一位必须为 0.
NRI: 2 个比特. nal_ref_idc. 取 00 ~ 11, 指示nalu单元的重要性，, 如 00 的 NALU 解码器可以丢弃而不影响图像的回放. 不过一般情况下不太关心这个属性.
Type: 5 个比特.nal_unit_type. 这个 NALU 单元的类型.
RTP打包原则
RTP的包长度必须要小于MTU(最大传输单元)，IP协议中MTU的最大长度为1500字节。除去IP报头（20字节）、UDP报头（8字节）、RTP头（12字节），所有RTP有效载荷（即NALU内容）的长度不得超过1460字节。TCP(20字节)。上面我的例子代码有参考，分包。
RTP有三种封包模式：单一封包模式，组合封包模式，分片封包模式
这是我的协议，我所有的通信协议，都会加上这么一个起始头，用来区别本系统，发送的数据包开始字段
3.4、PES分组头部
可以看到RTP数据头，协议，非常多，没什么难度，就是多，一般我们都不会从新造轮子，网上很多大佬，开源分享，移植过来，修改就OK了，下篇，直接贴代码，封装协议，要老命，自己去手撸出来，手撸linux内核代码没必要。
找到一个博主封装的h264，封包，解包
RTP荷载H264的代码参考：
http://blog.csdn.net/dengzikun/article/details/5807694
RTP荷载PS流的代码参考：
http://www.pudn.com/downloads33/sourcecode/windows/multimedia/detail105823.html
http://www.oschina.net/code/snippet_99626_23737
这是他说的
H264 RTP打包类、解包类，实现了单个NAL单元包和FU_A分片单元包。对于丢包处理，采用简单的策略：丢弃随后的所有数据包，直到收到关键帧。测试效果还不错，代码贴上来，若能为同道中人借鉴一二，足矣。两个类的使用说明如下(省略了错误处理过程)：
这是框架简介，封装的两个函数
DWORD H264SSRC ;CH264_RTP_PACK pack ( H264SSRC ) ;BYTE *pVideoData ;DWORD Size, ts ;bool IsEndOfFrame ;WORD wLen ;pack.Set ( pVideoData, Size, ts, IsEndOfFrame ) ;BYTE *pPacket ;while ( pPacket = pack.Get ( &wLen ) ){// rtp packet process// ...}HRESULT hr ;CH264_RTP_UNPACK unpack ( hr ) ;BYTE *pRtpData ;WORD inSize;int outSize ;BYTE *pFrame = unpack.Parse_RTP_Packet ( pRtpData, inSize, &outSize ) ;if ( pFrame != NULL ){// frame process// ...}
这是函数体
// class CH264_RTP_PACK start class CH264_RTP_PACK
{ #define RTP_VERSION 2 typedef struct NAL_msg_s { bool eoFrame ; unsigned char type; // NAL type unsigned char *start; // pointer to first location in the send buffer unsigned char *end; // pointer to last location in send buffer unsigned long size ; } NAL_MSG_t; typedef struct { //LITTLE_ENDIAN unsigned short cc:4; /* CSRC count */ unsigned short x:1; /* header extension flag */ unsigned short p:1; /* padding flag */ unsigned short v:2; /* packet type */ unsigned short pt:7; /* payload type */ unsigned short m:1; /* marker bit */ unsigned short seq; /* sequence number */ unsigned long ts; /* timestamp */ unsigned long ssrc; /* synchronization source */ } rtp_hdr_t; typedef struct tagRTP_INFO { NAL_MSG_t nal; // NAL information rtp_hdr_t rtp_hdr; // RTP header is assembled here int hdr_len; // length of RTP header unsigned char *pRTP; // pointer to where RTP packet has beem assembled unsigned char *start; // pointer to start of payload unsigned char *end; // pointer to end of payload unsigned int s_bit; // bit in the FU header unsigned int e_bit; // bit in the FU header bool FU_flag; // fragmented NAL Unit flag } RTP_INFO; public: CH264_RTP_PACK(unsigned long H264SSRC, unsigned char H264PAYLOADTYPE=96, unsigned short MAXRTPPACKSIZE=1472 ) { m_MAXRTPPACKSIZE = MAXRTPPACKSIZE ; if ( m_MAXRTPPACKSIZE > 10000 ) { m_MAXRTPPACKSIZE = 10000 ; } if ( m_MAXRTPPACKSIZE < 50 ) { m_MAXRTPPACKSIZE = 50 ; } memset ( &m_RTP_Info, 0, sizeof(m_RTP_Info) ) ; m_RTP_Info.rtp_hdr.pt = H264PAYLOADTYPE ; m_RTP_Info.rtp_hdr.ssrc = H264SSRC ; m_RTP_Info.rtp_hdr.v = RTP_VERSION ; m_RTP_Info.rtp_hdr.seq = 0 ; } ~CH264_RTP_PACK(void) { } //传入Set的数据必须是一个完整的NAL,起始码为0x00000001。 //起始码之前至少预留10个字节，以避免内存COPY操作。 //打包完成后，原缓冲区内的数据被破坏。 bool Set ( unsigned char *NAL_Buf, unsigned long NAL_Size, unsigned long Time_Stamp, bool End_Of_Frame ) { unsigned long startcode = StartCode(NAL_Buf) ; if ( startcode != 0x01000000 ) { return false ; } int type = NAL_Buf[4] & 0x1f ; if ( type < 1 || type > 12 ) { return false ; } m_RTP_Info.nal.start = NAL_Buf ; m_RTP_Info.nal.size = NAL_Size ; m_RTP_Info.nal.eoFrame = End_Of_Frame ; m_RTP_Info.nal.type = m_RTP_Info.nal.start[4] ; m_RTP_Info.nal.end = m_RTP_Info.nal.start + m_RTP_Info.nal.size ; m_RTP_Info.rtp_hdr.ts = Time_Stamp ; m_RTP_Info.nal.start += 4 ; // skip the syncword if ( (m_RTP_Info.nal.size + 7) > m_MAXRTPPACKSIZE ) { m_RTP_Info.FU_flag = true ; m_RTP_Info.s_bit = 1 ; m_RTP_Info.e_bit = 0 ; m_RTP_Info.nal.start += 1 ; // skip NAL header } else { m_RTP_Info.FU_flag = false ; m_RTP_Info.s_bit = m_RTP_Info.e_bit = 0 ; } m_RTP_Info.start = m_RTP_Info.end = m_RTP_Info.nal.start ; m_bBeginNAL = true ; return true ; } //循环调用Get获取RTP包，直到返回值为NULL unsigned char* Get ( unsigned short *pPacketSize ) { if ( m_RTP_Info.end == m_RTP_Info.nal.end ) { *pPacketSize = 0 ; return NULL ; } if ( m_bBeginNAL ) { m_bBeginNAL = false ; } else { m_RTP_Info.start = m_RTP_Info.end; // continue with the next RTP-FU packet } int bytesLeft = m_RTP_Info.nal.end - m_RTP_Info.start ; int maxSize = m_MAXRTPPACKSIZE - 12 ; // sizeof(basic rtp header) == 12 bytes if ( m_RTP_Info.FU_flag ) maxSize -= 2 ; if ( bytesLeft > maxSize ) { m_RTP_Info.end = m_RTP_Info.start + maxSize ; // limit RTP packetsize to 1472 bytes } else { m_RTP_Info.end = m_RTP_Info.start + bytesLeft ; } if ( m_RTP_Info.FU_flag ) { // multiple packet NAL slice if ( m_RTP_Info.end == m_RTP_Info.nal.end ) { m_RTP_Info.e_bit = 1 ; } } m_RTP_Info.rtp_hdr.m = m_RTP_Info.nal.eoFrame ? 1 : 0 ; // should be set at EofFrame if ( m_RTP_Info.FU_flag && !m_RTP_Info.e_bit ) { m_RTP_Info.rtp_hdr.m = 0 ; } m_RTP_Info.rtp_hdr.seq++ ; unsigned char *cp = m_RTP_Info.start ; cp -= ( m_RTP_Info.FU_flag ? 14 : 12 ) ; m_RTP_Info.pRTP = cp ; unsigned char *cp2 = (unsigned char *)&m_RTP_Info.rtp_hdr ; cp[0] = cp2[0] ; cp[1] = cp2[1] ; cp[2] = ( m_RTP_Info.rtp_hdr.seq >> 8 ) & 0xff ; cp[3] = m_RTP_Info.rtp_hdr.seq & 0xff ; cp[4] = ( m_RTP_Info.rtp_hdr.ts >> 24 ) & 0xff ; cp[5] = ( m_RTP_Info.rtp_hdr.ts >> 16 ) & 0xff ; cp[6] = ( m_RTP_Info.rtp_hdr.ts >> 8 ) & 0xff ; cp[7] = m_RTP_Info.rtp_hdr.ts & 0xff ; cp[8] = ( m_RTP_Info.rtp_hdr.ssrc >> 24 ) & 0xff ; cp[9] = ( m_RTP_Info.rtp_hdr.ssrc >> 16 ) & 0xff ; cp[10] = ( m_RTP_Info.rtp_hdr.ssrc >> 8 ) & 0xff ; cp[11] = m_RTP_Info.rtp_hdr.ssrc & 0xff ; m_RTP_Info.hdr_len = 12 ; /*! * /n The FU indicator octet has the following format: * /n * /n +---------------+ * /n MSB |0|1|2|3|4|5|6|7| LSB * /n +-+-+-+-+-+-+-+-+ * /n |F|NRI| Type | * /n +---------------+ * /n * /n The FU header has the following format: * /n * /n +---------------+ * /n |0|1|2|3|4|5|6|7| * /n +-+-+-+-+-+-+-+-+ * /n |S|E|R| Type | * /n +---------------+ */ if ( m_RTP_Info.FU_flag ) { // FU indicator F|NRI|Type cp[12] = ( m_RTP_Info.nal.type & 0xe0 ) | 28 ; //Type is 28 for FU_A //FU header S|E|R|Type cp[13] = ( m_RTP_Info.s_bit << 7 ) | ( m_RTP_Info.e_bit << 6 ) | ( m_RTP_Info.nal.type & 0x1f ) ; //R = 0, must be ignored by receiver m_RTP_Info.s_bit = m_RTP_Info.e_bit= 0 ; m_RTP_Info.hdr_len = 14 ; } m_RTP_Info.start = &cp[m_RTP_Info.hdr_len] ; // new start of payload *pPacketSize = m_RTP_Info.hdr_len + ( m_RTP_Info.end - m_RTP_Info.start ) ; return m_RTP_Info.pRTP ; } private: unsigned int StartCode( unsigned char *cp ) { unsigned int d32 ; d32 = cp[3] ; d32 <<= 8 ; d32 |= cp[2] ; d32 <<= 8 ; d32 |= cp[1] ; d32 <<= 8 ; d32 |= cp[0] ; return d32 ; } private: RTP_INFO m_RTP_Info ; bool m_bBeginNAL ; unsigned short m_MAXRTPPACKSIZE ;
}; // class CH264_RTP_PACK end
//
// class CH264_RTP_UNPACK start class CH264_RTP_UNPACK
{ #define RTP_VERSION 2
#define BUF_SIZE (1024 * 500) typedef struct { //LITTLE_ENDIAN unsigned short cc:4; /* CSRC count */ unsigned short x:1; /* header extension flag */ unsigned short p:1; /* padding flag */ unsigned short v:2; /* packet type */ unsigned short pt:7; /* payload type */ unsigned short m:1; /* marker bit */ unsigned short seq; /* sequence number */ unsigned long ts; /* timestamp */ unsigned long ssrc; /* synchronization source */ } rtp_hdr_t;
public: CH264_RTP_UNPACK ( HRESULT &hr, unsigned char H264PAYLOADTYPE = 96 ) : m_bSPSFound(false) , m_bWaitKeyFrame(true) , m_bPrevFrameEnd(false) , m_bAssemblingFrame(false) , m_wSeq(1234) , m_ssrc(0) { m_pBuf = new BYTE[BUF_SIZE] ; if ( m_pBuf == NULL ) { hr = E_OUTOFMEMORY ; return ; } m_H264PAYLOADTYPE = H264PAYLOADTYPE ; m_pEnd = m_pBuf + BUF_SIZE ; m_pStart = m_pBuf ; m_dwSize = 0 ; hr = S_OK ; } ~CH264_RTP_UNPACK(void) { delete [] m_pBuf ; } //pBuf为H264 RTP视频数据包，nSize为RTP视频数据包字节长度，outSize为输出视频数据帧字节长度。 //返回值为指向视频数据帧的指针。输入数据可能被破坏。 BYTE* Parse_RTP_Packet ( BYTE *pBuf, unsigned short nSize, int *outSize ) { if ( nSize <= 12 ) { return NULL ; } BYTE *cp = (BYTE*)&m_RTP_Header ; cp[0] = pBuf[0] ; cp[1] = pBuf[1] ; m_RTP_Header.seq = pBuf[2] ; m_RTP_Header.seq <<= 8 ; m_RTP_Header.seq |= pBuf[3] ; m_RTP_Header.ts = pBuf[4] ; m_RTP_Header.ts <<= 8 ; m_RTP_Header.ts |= pBuf[5] ; m_RTP_Header.ts <<= 8 ; m_RTP_Header.ts |= pBuf[6] ; m_RTP_Header.ts <<= 8 ; m_RTP_Header.ts |= pBuf[7] ; m_RTP_Header.ssrc = pBuf[8] ; m_RTP_Header.ssrc <<= 8 ; m_RTP_Header.ssrc |= pBuf[9] ; m_RTP_Header.ssrc <<= 8 ; m_RTP_Header.ssrc |= pBuf[10] ; m_RTP_Header.ssrc <<= 8 ; m_RTP_Header.ssrc |= pBuf[11] ; BYTE *pPayload = pBuf + 12 ; DWORD PayloadSize = nSize - 12 ; // Check the RTP version number (it should be 2): if ( m_RTP_Header.v != RTP_VERSION ) { return NULL ; } /* // Skip over any CSRC identifiers in the header: if ( m_RTP_Header.cc ) { long cc = m_RTP_Header.cc * 4 ; if ( Size < cc ) { return NULL ; } Size -= cc ; p += cc ; } // Check for (& ignore) any RTP header extension if ( m_RTP_Header.x ) { if ( Size < 4 ) { return NULL ; } Size -= 4 ; p += 2 ; long l = p[0] ; l <<= 8 ; l |= p[1] ; p += 2 ; l *= 4 ; if ( Size < l ) ; { return NULL ; } Size -= l ; p += l ; } // Discard any padding bytes: if ( m_RTP_Header.p ) { if ( Size == 0 ) { return NULL ; } long Padding = p[Size-1] ; if ( Size < Padding ) { return NULL ; } Size -= Padding ; }*/ // Check the Payload Type. if ( m_RTP_Header.pt != m_H264PAYLOADTYPE ) { return NULL ; } int PayloadType = pPayload[0] & 0x1f ; int NALType = PayloadType ; if ( NALType == 28 ) // FU_A { if ( PayloadSize < 2 ) { return NULL ; } NALType = pPayload[1] & 0x1f ; } if ( m_ssrc != m_RTP_Header.ssrc ) { m_ssrc = m_RTP_Header.ssrc ; SetLostPacket () ; } if ( NALType == 0x07 ) // SPS { m_bSPSFound = true ; } if ( !m_bSPSFound ) { return NULL ; } if ( NALType == 0x07 || NALType == 0x08 ) // SPS PPS { m_wSeq = m_RTP_Header.seq ; m_bPrevFrameEnd = true ; pPayload -= 4 ; *((DWORD*)(pPayload)) = 0x01000000 ; *outSize = PayloadSize + 4 ; return pPayload ; } if ( m_bWaitKeyFrame ) { if ( m_RTP_Header.m ) // frame end { m_bPrevFrameEnd = true ; if ( !m_bAssemblingFrame ) { m_wSeq = m_RTP_Header.seq ; return NULL ; } } if ( !m_bPrevFrameEnd ) { m_wSeq = m_RTP_Header.seq ; return NULL ; } else { if ( NALType != 0x05 ) // KEY FRAME { m_wSeq = m_RTP_Header.seq ; m_bPrevFrameEnd = false ; return NULL ; } } } /// if ( m_RTP_Header.seq != (WORD)( m_wSeq + 1 ) ) // lost packet { m_wSeq = m_RTP_Header.seq ; SetLostPacket () ; return NULL ; } else { // 码流正常 m_wSeq = m_RTP_Header.seq ; m_bAssemblingFrame = true ; if ( PayloadType != 28 ) // whole NAL { *((DWORD*)(m_pStart)) = 0x01000000 ; m_pStart += 4 ; m_dwSize += 4 ; } else // FU_A { if ( pPayload[1] & 0x80 ) // FU_A start { *((DWORD*)(m_pStart)) = 0x01000000 ; m_pStart += 4 ; m_dwSize += 4 ; pPayload[1] = ( pPayload[0] & 0xE0 ) | NALType ; pPayload += 1 ; PayloadSize -= 1 ; } else { pPayload += 2 ; PayloadSize -= 2 ; } } if ( m_pStart + PayloadSize < m_pEnd ) { CopyMemory ( m_pStart, pPayload, PayloadSize ) ; m_dwSize += PayloadSize ; m_pStart += PayloadSize ; } else // memory overflow { SetLostPacket () ; return NULL ; } if ( m_RTP_Header.m ) // frame end { *outSize = m_dwSize ; m_pStart = m_pBuf ; m_dwSize = 0 ; if ( NALType == 0x05 ) // KEY FRAME { m_bWaitKeyFrame = false ; } return m_pBuf ; } else { return NULL ; } } } void SetLostPacket() { m_bSPSFound = false ; m_bWaitKeyFrame = true ; m_bPrevFrameEnd = false ; m_bAssemblingFrame = false ; m_pStart = m_pBuf ; m_dwSize = 0 ; } private: rtp_hdr_t m_RTP_Header ; BYTE *m_pBuf ; bool m_bSPSFound ; bool m_bWaitKeyFrame ; bool m_bAssemblingFrame ; bool m_bPrevFrameEnd ; BYTE *m_pStart ; BYTE *m_pEnd ; DWORD m_dwSize ; WORD m_wSeq ; BYTE m_H264PAYLOADTYPE ; DWORD m_ssrc ;
}; // class CH264_RTP_UNPACK end
四、推拉流测试
另一个博主封装了一个推拉流的demo，可以移植到项目中，对h264，aac，感谢博主的热心分享。
他用FFmpeg去对一个事先准备好的mp4文件，读取流，然后通过RTSP协议，推流到一个文件夹，然后写了个客户端，rtsp，拉流，播放。
如果我么要移植到ipc项目中，这里需要修改一下，将soc采集到的视频流，放到rtp包里，去掉FFmpeg解码，也不需要移植FFmpeg。
原文链接：https://blog.csdn.net/weixin_43147845/article/details/140923649
地址：https://github.com/BreakingY/simple-rtsp-client
1、准备
simple-rtsp-server依赖ffmpeg，版本要求>=4.x。支持系统：Linux
依赖安装：
sudo apt-get -y install autoconf automake build-essential libass-dev libfreetype6-dev libsdl2-dev libtheora-dev libtool libva-dev libvdpau-dev libvorbis-dev libxcb1-dev libxcb-shm0-dev libxcb-xfixes0-dev pkg-config texinfo zlib1g-dev
汇编库：
sudo apt-get install yasm
sudo apt-get install nasm
视频库：
sudo apt-get install libx264-dev
sudo apt-get install libx265-dev
音频库：
sudo apt-get install libfdk-aac-dev
sudo apt-get install libmp3lame-dev
sudo apt-get install libopus-dev
ffmpeg源码下载：
wget https://ffmpeg.org//releases/ffmpeg-4.0.5.tar.bz2
tar xjvf ffmpeg-4.0.5.tar.bz2
cd ffmpeg-4.0.5
编译安装：
./configure --prefix=/usr/local --enable-libx264 --disable-x86asm --enable-nonfree --enable-libfdk-aac --enable-shared --enable-gpl --enable-libmp3lame --enable-libopus --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/libmakemake install
2、simple-rtsp-server下载编译
下载后
cd simple-rtsp-servermkdir buildcd buildcmake ..make -j
3、运行
cp -r ../mp4path ../rtsp_server 0 (0-不鉴权；1-鉴权)
4、拉流测试
项目中mp4path自带了测试文件，后面把想回放的视频放到mp4path中即可
TCP拉流：
ffmpeg -rtsp_transport tcp -i "rtsp://192.168.10.17:8554/test_h264_aac.mp4" -vcodec copy -acodec copy test_h264_aac_tcp.mp4
UDP拉流：
ffmpeg -i "rtsp://192.168.10.17:8554/test_h264_aac.mp4" -vcodec copy -acodec copy test_h264_aac_udp.mp4
也可通过VLC直接播放，点击媒体->打开网络串流，输入rtsp地址即可。默认是udp拉流，要使用TCP需要打开工具->偏好设置->输入/编解码器，拉到最下方，选择“RTP over RTSP（TCP）”
他也写了一个客户端拉流，不用上面测试命令FFmpeg
二、RTSP Client实战项目
地址：https://github.com/BreakingY/simple-rtsp-client
支持RTP OVER UDP、RTP OVER TCP，支持H264/H265、AAC/PCMA、支持鉴权。
不需要任何依赖。
1、下载后编译
mkdir buildcd buildcmake ..make -j
2、测试
./rtsp_client rtsp_url
客户端会把收到的音视频写入文件，H264/H265写入到test_out.h26x，AAC写入到test_out.aac，PCMA写入到test_out.pcma。
另一个比较厉害的，比较全
https://github.com/ireader/media-server
