“RTMP study notes”的版本间的差异

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第88行： 第88行：
 
! ID      !! Name          !! Description   
 
! ID      !! Name          !! Description   
 
|-  
 
|-  
−
||17(AMF3),20(AMF0)|| Command Message     || These messages are sent to perform some operations like
+
||17(AMF3),20(AMF0)|| Command Message**    || These messages are sent to perform some operations like
 
connect, createStream, publish, play, pause on the peer. Command
 
connect, createStream, publish, play, pause on the peer. Command
 
messages like onstatus, result etc. are used to inform the sender
 
messages like onstatus, result etc. are used to inform the sender
第105行： 第105行：
  
 
|}
 
|}
 +
*Limit type
 +
0 - Hard: The peer SHOULD limit its output bandwidth to the indicated window size.
 +
1 - Soft: The peer SHOULD limit its output bandwidth to the the window indicated in this message or the limit already in effect, whichever is smaller.
 +
2 - Dynamic: If the previous Limit Type was Hard, treat this message as though it was marked Hard, otherwise ignore this message.
 +
 +
*RTMP Message Format
 +
Message Header: big-endian format
 +
Message Type[1]
 +
Payload length[3]
 +
Timestamp[4]
 +
Stream ID [3]
 +
 +
Message payload: actual data
  
−
*User Control Message - Event Type
+
**User Control Message - Event Type
 
{| class="wikitable" style="width:100%"
 
{| class="wikitable" style="width:100%"
 
! ID      !! Event          !!Event data!! Description   
 
! ID      !! Event          !!Event data!! Description   
第126行： 第139行：
 
|}
 
|}
  
−
*Limit type
 
−
0 - Hard: The peer SHOULD limit its output bandwidth to the indicated window size.
 
−
1 - Soft: The peer SHOULD limit its output bandwidth to the the window indicated in this message or the limit already in effect, whichever is smaller.
 
−
2 - Dynamic: If the previous Limit Type was Hard, treat this message as though it was marked Hard, otherwise ignore this message.
 
  
−
*RTMP Message Format
+
**Types of command
−
Message Header: big-endian format
+
1. NetConnection
−
Message Type[1]
+
{| class="wikitable" style="width:100%"
−
  Payload length[3]
+
! command    !! description          !! fields
−
  Timestamp[4]
+
|-  
−
  Stream ID [3]
+
|| connect    || client: request connect    || command_name+Transaction ID+commans object+user argument
 +
|-  
 +
|| call || remote procedure calls || procedureName + Tr. ID+commans object+ argument
 +
|-  
 +
|| createStream|| client: request server to create a logical channel for message || procedureName + Tr. ID+commans object
 +
|}
  
−
Message payload: actual data
+
Note:
 +
Audio supported: SUPPORT_SND_ADPCM, MP3, NELLY8, NELLY, G711A, G711U, NELLY16,AAC, SPEEX
 +
Video supported: SUPPORT_VID_SORENSON, HOMEBREW, VP6, VP6ALPHA, HOMEBREWV2, H264
 +
AMF0: AMF0 object encoding supported by Flash 6 and later
 +
AMF3: AMF3 encoding from Flash 9 (AS3)
 +
 
 +
2. NetStream
 +
1. NetConnection
 +
{| class="wikitable" style="width:100%"
 +
! command    !! description          !! fields
 +
|-
 +
|| play|| client:      || command_name+Transaction ID+commans object+user argument
 +
|- 
 +
|| deleteStream||client:      || 
 +
|- 
 +
|| closeStream|| client: ||
 +
|- 
 +
|| receiveAudio|| client: ||
 +
|- 
 +
|| reveiveVideo|| client: ||
 +
|- 
 +
|| publish|| client: ||
 +
|- 
 +
||seek|| client: ||
 +
|- 
 +
||pause|| client: || The client sends the pause command to tell the server to pause or
 +
start playing
 +
|- 
 +
||onStatus|| server: status update|| 
 +
|}
 +
= examples =
 +
Use nginx as server and ffmpeg as clients
 +
Ref: cnblogs.com/liangblog/p/11122237.html
 +
 
 +
*source download:
 +
nginx:
 +
http://nginx.org/download/nginx-1.16.1.tar.gz
 +
wget https://github.com/arut/nginx-rtmp-module/archive/master.zip
 +
 
 +
openssl:
 +
openssl-1.1.1
 +
openssl-3.0.2
 +
rtmpdump:
 +
git clone https://git.ffmpeg.org/rtmpdump
 +
 
 +
ffmpeg:
 +
git clone https://git.ffmpeg.org/ffmpeg.git ffmpeg
 +
 
 +
*build
 +
nginx:
 +
./configure --add-module=nginx-rtmp-module
 +
make
 +
sudo make install
 +
<source lang="C">
 +
  nginx path prefix: "/usr/local/nginx"
 +
  nginx binary file: "/usr/local/nginx/sbin/nginx"
 +
  nginx modules path: "/usr/local/nginx/modules"
 +
  nginx configuration prefix: "/usr/local/nginx/conf"
 +
  nginx configuration file: "/usr/local/nginx/conf/nginx.conf"
 +
  nginx pid file: "/usr/local/nginx/logs/nginx.pid"
 +
  nginx error log file: "/usr/local/nginx/logs/error.log"
 +
  nginx http access log file: "/usr/local/nginx/logs/access.log"
 +
  nginx http client request body temporary files: "client_body_temp"
 +
  nginx http proxy temporary files: "proxy_temp"
 +
  nginx http fastcgi temporary files: "fastcgi_temp"
 +
  nginx http uwsgi temporary files: "uwsgi_temp"
 +
  nginx http scgi temporary files: "scgi_temp"
 +
</source>
 +
 
 +
Nginx会被安装在/usr/local/nginx目录下
 +
vi /usr/local/nginx/conf/nginx.conf
 +
<source lang="c">
 +
events  {
 +
  //events模块来用指定nginx的工作模式和工作模式及连接数上限
 +
}
 +
http        {
 +
  //负责HTTP服务器相关属性的配置
 +
}
 +
rtmp {
 +
    server {
 +
        listen 1935;
 +
        buflen 10s;
 +
        application myapp {
 +
            live on;
 +
            max_connections 1024;
 +
        }
 +
        application mgclient {
 +
            live on;
 +
            hls on;
 +
            hls_fragment_naming sequential;
 +
            hls_playlist_length 30s;
 +
            hls_continuous on;
 +
 
 +
        }
 +
    }
 +
}
 +
 
 +
</source>
 +
 
 +
rtmpdump:
 +
make sys=posix
 +
sudo make install
 +
 
 +
ffmpeg:
 +
prerequirements: sudo apt install pkg-config
 +
./configure --enable-librtmp
 +
make
 +
sudo make install
 +
 
 +
 
 +
*start the server
 +
启动      ：/usr/local/nginx/sbin/nginx
 +
检查配置文件：/usr/local/nginx/sbin/nginx -t
 +
重载配置文件：/usr/local/nginx/sbin/nginx -s reload
 +
重启      : /usr/local/nginx/sbin/nginx -s reopen
 +
停止      ：/usr/local/nginx/sbin/nginx -s stop
 +
 
 +
*push a mp4 file
 +
ffmpeg -re -i ${HOME}/test.mp4 -vcodec copy -codec copy -f flv rtmp://localhost/myapp/test1
 +
[[文件:push.jpg]]
 +
*play a mp4
 +
ffplay rtmp://localhost/myapp/test1
 +
[[文件:pull.jpg]]
 +
 
 +
[[*libRTMP study]]
 +
 
 +
=Trouble shootings=
 +
1. prerequirements:
 +
  openssl library: apt install libssl-dev && sudo apt install OpenSSL
 +
  PCRE library: no
 +
  zlib: sudo apt install zlib1g-dev
 +
  nginx configure:
 +
  ./configure --prefix=/usr/local/nginx --add-module=./nginx-rtmp-module --with-http_ssl_module --without-http_rewrite_module
 +
2. openssl 3.x compiling error
 +
  vi nginx/objs/Makefile => CFLAGS += -Wno-warning -Wno-deprecated-declarations
 +
 
 +
3. fix librtmp compiling errors:
 +
librtmp/dh.h
 +
<source lang="c">
 +
typedef struct dh_st {
 +
  MP_t p;
 +
  MP_t g;
 +
  MP_t pub_key;
 +
  MP_t priv_key;
 +
  long length;
 +
} MDH;
 +
</source>
 +
librtmp/handshake.h,
 +
<source lang="c">
 +
#define HMAC_setup(ctx, key, len) { ctx=HMAC_CTX_new(); HMAC_Init_ex(ctx, key, len, EVP_sha256(), 0);}
 +
#define HMAC_crunch(ctx, buf, len) HMAC_Update(ctx, buf, len)
 +
#define HMAC_finish(ctx, dig, dlen) HMAC_Final(ctx, dig, &dlen); HMAC_CTX_free(ctx)
 +
 
 +
static void HMACsha256(const uint8_t *message, size_t messageLen, ...)
 +
static void InitRC4Encryption() {
 +
    HMAC_CTX* ctx;
 +
    HMAC_setup(ctx, secretKey, 128);
 +
    HMAC_crunch(ctx, message, messageLen);
 +
    HMAC_finish(ctx, digest, digestLen);
 +
 
 +
}
 +
 
 +
</source>
 +
librtmp/hashswf.c
 +
<source lang="c">
 +
struct info
 +
{
 +
  z_stream *zs;
 +
  HMAC_CTX* ctx;
 +
  int first;
 +
  int zlib;
 +
  int size;
 +
};
 +
 
 +
#else  /* USE_OPENSSL */
 +
#include <openssl/ssl.h>
 +
#include <openssl/sha.h>
 +
#include <openssl/hmac.h>
 +
#include <openssl/rc4.h>
 +
#define HMAC_setup(ctx, key, len)      { ctx=HMAC_CTX_new(); HMAC_Init_ex(ctx, (unsigned char *)key, len, EVP_sha256(), 0);}
 +
#define HMAC_crunch(ctx, buf, len)      HMAC_Update(ctx, (unsigned char *)buf, len)
 +
#define HMAC_finish(ctx, dig, dlen)    HMAC_Final(ctx, (unsigned char *)dig, &dlen);
 +
#define HMAC_close(ctx) HMAC_CTX_free(ctx)
 +
#endif
 +
 
 +
</source>
 +
 
 +
4. For new OS, you will need
 +
- X11: sudo apt install xorg-dev
 +
- ALSA: sudo apt install libpulse-dev
 +
- SDL: ./configure --enable-video-x11 --enable-x11-shared --enable-video-x11-vm --enable-alsa

2022年5月4日 (三) 13:19的最后版本

Ref: https://www.adobe.com/devnet/rtmp.html
Handshaking Sequence

uninitialized ----

C0 -->
C1 -->
<-- S0 必须在 C0或C1 之后
<-- S1 必须在 C0或C1 之后

version sent -----

C2 -->
C2 必须在 S1之后

Ack sent -----

<-- s2 必须在C1之后

Handshake Done ----

data -->
data必须在S2之后 

  <-- data 必须在c2 之后

C0-> RTMP version 8 bits (=3) 

 <- S0 RTMP version 8 bit (=3)

C1,S1 1536 bytes 

time[4]:   stream 的参考基准时间
zero[4]:0
random bytes[1528]  乱数token

C2,S2 1536 bytes 

time[4]:   C2: S1 的时间; S2: C1 的时间
time2[4]:  读取S1/C1的时间
random bytes[1528]: C1/S1的乱数token

Chunk:
| Basic Header | Message Header | Extended Timestamp | Chunk Data |
+--------------+----------------+--------------------+--------------+
Basic Header: 1~3 bytes, Chunk Stream ID + chunk type
Message Header:0,3,7, or 11 bytes,message的type
Extended Timestamp: 0 or 4 bytes
Chunk header = Basic Header + Message Header + Extended Timestamp
Chunk data: variable size.
Chunk Stream ID: 3~65599, 

 0n = 64+n (64~319)
 1mn = 64+m + n*256 (64~65599)
 n (n !=0,1,2 one byte) (3~63)
 2 is reserved for level control

chunk type 高2位:format of Message header 

Type 0: 11 byte 用于stream 开头
 timestamp[3]: 发送时间, 0xffffff  使用Extended Timestamp [32bits]
 message length[3]
 message type id[1]
 message stream id[4]
Type 1: 7 bytes
 timestamp delta[3]
 message length[3]
 message type id[1]
Type 2: 3 bytes ID 与长度跟前个chunk 同
 timestamp delta[3]
Type 3: no message header. timestamp , ID 与长度跟前个chunk 同

message type id 

1,2,3,5,6 for protocol control message
8: audio
9: video

Protocol Control Messages
message stream ID 0: control stream
chunk stream ID 2: used in control stream
message type id

ID Name Fields size Meaning
1 Set Chunk Size maximum chunk size 32 bits Set maximum chunk size (1~0x7FFFFFFF)
2 Abort Message chunk stream ID 32 bits chunk stream ID message to be discard
3 Acknowledgement sequnce number 32 bits This field holds the number of bytes

received so far.

5 Window Acknowledgement Size window size 32 bits inform the peer of the

window size to use between sending acknowledgments

6 Set Peer Bandwidth Acknowledgement Window size+Limit Type 32+8 sends this message to limit the output
4 User Control Message Event Type*+Event Data 16+variable contain information used by the RTMP streaming layer

Message Type:

ID Name Description
17(AMF3),20(AMF0) Command Message** These messages are sent to perform some operations like

connect, createStream, publish, play, pause on the peer. Command
messages like onstatus, result etc. are used to inform the sender
about the status of the requested commands.

18(AMF0),15(AMF3) Data Message metadata or user data
19(AMF0),16(AMF3) Share Object Message Flash object: Shared object Name+Current version+Flags+Event Type+Event data length+Event data
8 Audio Message audio data
9 Video Message video data
22 Aggregate Message contains a series of RTMP sub-message


  • Limit type

0 - Hard: The peer SHOULD limit its output bandwidth to the indicated window size.
1 - Soft: The peer SHOULD limit its output bandwidth to the the window indicated in this message or the limit already in effect, whichever is smaller.
2 - Dynamic: If the previous Limit Type was Hard, treat this message as though it was marked Hard, otherwise ignore this message.

  • RTMP Message Format

Message Header: big-endian format 

Message Type[1]
Payload length[3]
Timestamp[4]
Stream ID [3]

Message payload: actual data

    • User Control Message - Event Type
ID Event Event data Description
0 Stream Begin stream ID[4] The server sends this event to notify the client that a stream has become functional
1 Stream EOF stream ID[4] server: playback of data is over
2 StreamDry stream ID[4] server: no more data on this stream
3 SetBufferLenth stream ID[4]+bufferLength_in_ms[4] client:inform server the buffer size in miliseconds
4 StreamIsRecorded stream ID[4] Server: stream is recorded stream
6 PingRequest timestamp[4] Server: test whether client is reachable
7 Ping Response timestamp[4] Client: responseserver with 6's timestamp


    • Types of command

1. NetConnection

command description fields
connect client: request connect command_name+Transaction ID+commans object+user argument
call remote procedure calls procedureName + Tr. ID+commans object+ argument
createStream client: request server to create a logical channel for message procedureName + Tr. ID+commans object

Note:
Audio supported: SUPPORT_SND_ADPCM, MP3, NELLY8, NELLY, G711A, G711U, NELLY16,AAC, SPEEX
Video supported: SUPPORT_VID_SORENSON, HOMEBREW, VP6, VP6ALPHA, HOMEBREWV2, H264
AMF0: AMF0 object encoding supported by Flash 6 and later
AMF3: AMF3 encoding from Flash 9 (AS3)

2. NetStream
1. NetConnection

command description fields
play client: command_name+Transaction ID+commans object+user argument
deleteStream client:
closeStream client:
receiveAudio client:
reveiveVideo client:
publish client:
seek client:
pause client: The client sends the pause command to tell the server to pause or

start playing

onStatus server: status update

examples

Use nginx as server and ffmpeg as clients
Ref: cnblogs.com/liangblog/p/11122237.html

  • source download:

nginx: 

http://nginx.org/download/nginx-1.16.1.tar.gz
wget https://github.com/arut/nginx-rtmp-module/archive/master.zip

openssl: 

openssl-1.1.1
openssl-3.0.2

rtmpdump: 

git clone https://git.ffmpeg.org/rtmpdump

ffmpeg: 

git clone https://git.ffmpeg.org/ffmpeg.git ffmpeg
  • build

nginx: 

./configure --add-module=nginx-rtmp-module
make
sudo make install

  nginx path prefix: "/usr/local/nginx"
  nginx binary file: "/usr/local/nginx/sbin/nginx"
  nginx modules path: "/usr/local/nginx/modules"
  nginx configuration prefix: "/usr/local/nginx/conf"
  nginx configuration file: "/usr/local/nginx/conf/nginx.conf"
  nginx pid file: "/usr/local/nginx/logs/nginx.pid"
  nginx error log file: "/usr/local/nginx/logs/error.log"
  nginx http access log file: "/usr/local/nginx/logs/access.log"
  nginx http client request body temporary files: "client_body_temp"
  nginx http proxy temporary files: "proxy_temp"
  nginx http fastcgi temporary files: "fastcgi_temp"
  nginx http uwsgi temporary files: "uwsgi_temp"
  nginx http scgi temporary files: "scgi_temp"

Nginx会被安装在/usr/local/nginx目录下
vi /usr/local/nginx/conf/nginx.conf 

events   {
  //events模块来用指定nginx的工作模式和工作模式及连接数上限
}
http        {
  //负责HTTP服务器相关属性的配置
}
rtmp {
    server {
        listen 1935;
        buflen 10s;
        application myapp {
            live on;
            max_connections 1024;
        }
        application mgclient {
            live on;
            hls on;
            hls_fragment_naming sequential;
            hls_playlist_length 30s;
            hls_continuous on;
 
        }
    }
}

rtmpdump: 

make sys=posix
sudo make install

ffmpeg: 

prerequirements: sudo apt install pkg-config
./configure --enable-librtmp
make
sudo make install


  • start the server 
启动      ：/usr/local/nginx/sbin/nginx
检查配置文件：/usr/local/nginx/sbin/nginx -t
重载配置文件：/usr/local/nginx/sbin/nginx -s reload
重启      : /usr/local/nginx/sbin/nginx -s reopen
停止      ：/usr/local/nginx/sbin/nginx -s stop
  • push a mp4 file 
ffmpeg -re -i ${HOME}/test.mp4 -vcodec copy -codec copy -f flv rtmp://localhost/myapp/test1

Push.jpg

  • play a mp4 
ffplay rtmp://localhost/myapp/test1

Pull.jpg

*libRTMP study

Trouble shootings

1. prerequirements: 

 openssl library: apt install libssl-dev && sudo apt install OpenSSL
 PCRE library: no
 zlib: sudo apt install zlib1g-dev
 nginx configure:
 ./configure --prefix=/usr/local/nginx --add-module=./nginx-rtmp-module --with-http_ssl_module --without-http_rewrite_module

2. openssl 3.x compiling error 

 vi nginx/objs/Makefile => CFLAGS += -Wno-warning -Wno-deprecated-declarations

3. fix librtmp compiling errors:
librtmp/dh.h 

typedef struct dh_st {
  MP_t p;
  MP_t g;
  MP_t pub_key;
  MP_t priv_key;
  long length;
} MDH;

librtmp/handshake.h, 

#define HMAC_setup(ctx, key, len) { ctx=HMAC_CTX_new(); HMAC_Init_ex(ctx, key, len, EVP_sha256(), 0);}
#define HMAC_crunch(ctx, buf, len) HMAC_Update(ctx, buf, len)
#define HMAC_finish(ctx, dig, dlen) HMAC_Final(ctx, dig, &dlen); HMAC_CTX_free(ctx)
 
static void HMACsha256(const uint8_t *message, size_t messageLen, ...)
static void InitRC4Encryption() {
    HMAC_CTX* ctx;
    HMAC_setup(ctx, secretKey, 128);
    HMAC_crunch(ctx, message, messageLen);
    HMAC_finish(ctx, digest, digestLen);
 
}

librtmp/hashswf.c 

struct info
{
  z_stream *zs;
  HMAC_CTX* ctx;
  int first;
  int zlib;
  int size;
};
 
#else   /* USE_OPENSSL */
#include <openssl/ssl.h>
#include <openssl/sha.h>
#include <openssl/hmac.h>
#include <openssl/rc4.h>
#define HMAC_setup(ctx, key, len)       { ctx=HMAC_CTX_new(); HMAC_Init_ex(ctx, (unsigned char *)key, len, EVP_sha256(), 0);}
#define HMAC_crunch(ctx, buf, len)      HMAC_Update(ctx, (unsigned char *)buf, len)
#define HMAC_finish(ctx, dig, dlen)     HMAC_Final(ctx, (unsigned char *)dig, &dlen);
#define HMAC_close(ctx) HMAC_CTX_free(ctx)
#endif

4. For new OS, you will need 

- X11: sudo apt install xorg-dev
- ALSA: sudo apt install libpulse-dev
- SDL: ./configure --enable-video-x11 --enable-x11-shared --enable-video-x11-vm --enable-alsa
来自“https://www.mywiki.cn/free6d1823/index.php?title=RTMP_study_notes&oldid=88805”