User manual

（RTSP Server）

HAPPYTIME SOFTWARE CO., LIMITED

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www.happytimesoft.com
 Table of Contents
Chapter 1 Introduction .................................................................................................................. 4
Chapter 2 Key features .................................................................................................................. 5
Chapter 3 Function chart .............................................................................................................. 6
Chapter 4 Configuration ............................................................................................................... 7
4.1 Configuration Templates ..................................................................................................... 7
4.2 Configuring Node Description .......................................................................................... 10
4.2.1 System parameters ................................................................................................. 10
4.2.2 http_notify .............................................................................................................. 12
4.2.3 User node ............................................................................................................... 13
4.2.4 Output node ............................................................................................................ 14
4.2.5 Proxy node.............................................................................................................. 16
4.2.6 Pusher node............................................................................................................ 17
4.2.7 Backchannel node .................................................................................................. 18
Chapter 5 Data pusher................................................................................................................. 20
Chapter 6 RTSP over HTTP ....................................................................................................... 22
Chapter 7 RTSP over Websocket ............................................................................................... 24
Chapter 8 RTP Multicast ............................................................................................................ 25
Chapter 9 Audio back channel.................................................................................................... 26
9.1 RTSP Require- Tag ........................................................................................................... 26
9.2 Connection setup for a bi- directional connection............................................................ 26
9.3 Example ........................................................................................................................... 29
Chapter 10 Run RTSP Server ..................................................................................................... 30
Chapter 11 Multiple capture devices support ........................................................................... 31
Chapter 12 Capture application window ................................................................................... 32
Chapter 13 Support URL parameters ........................................................................................ 33
Chapter 14 Support SRTP .......................................................................................................... 35
14.1 SRTP for publishing ........................................................................................................ 35
14.2 SRTP for playback .......................................................................................................... 36
 Chapter 1 Introduction
Happytime RTSP Server is a complete RTSP server application. It can stream audio
and video files in various formats.
It can also stream video from camera, living screen and application windows,
stream audio from audio device.
It can stream H265, H264, MP4, MJPEG video stream and G711, G722, G726, AAC,
OPUS audio stream.
These streams can be received/played by standards-compliant RTSP/RTP media
clients.
It supports rtsp proxy function.
It supports audio back channel function.
It supports rtsp over http function.
It supports rtsp over https function.
It supports rtp multicast function.
Support for data pusher function.
Enjoying multimedia content from your computer can be a pleasant way for you
to spend your free time. However, sometimes you might need to access it from various
locations, such as a different computer or a handheld device, Happytime RTSP Server,
that can help you achieve quick and efficient results.
 Chapter 2 Key features
The server can transmit multiple streams concurrently
It can stream audio and video files in various formats
It can stream audio from audio device
It can stream video from camera and living screen
It can stream video from application windows
It can stream H265, H264, MP4, MJPEG video stream
It can stream G711, G722, G726, AAC, OPUS audio stream
It supports rtsp over http function
It supports rtsp over https function
It supports rtsp over websocket function
It supports rtp multicast function
It supports data pusher function
It supports srtp
It supports RTSP proxy function, as the following:

It supports audio backchannel

Happytime rtsp server complies with onvif audio backchannel
specification,please refer to the link below for specification details:
https://happytimesoft.com/knowledge/audio-back-channel.html
It supports audio and video playback.
Happytime rtsp server complies with onvif audio and video playback
specification,please refer to the link below for specification details:
https://happytimesoft.com/knowledge/audio-video-playback.html
 Chapter 3 Function chart
The function diagram of Happytime rtsp sever is as follows:
 Chapter 4 Configuration
If no configuration file is specified at startup, the default configuration file
rtspserver.cfg will be used.

4.1 Configuration Templates

<?xml version="1.0" encoding="utf-8"?>
<config>
<serverip></serverip>
<serverport>554</serverport>
<loop_nums>1</loop_nums>
<multicast>0</multicast>
<udp_base_port>22000</udp_base_port>
<metadata>1</metadata>
<rtsp_over_http>1</rtsp_over_http>
<http_port>8080</http_port>
<rtsp_over_https>1</rtsp_over_https>
<https_port>443</https_port>
<cert_file>ssl.ca</cert_file>
<key_file>ssl.key</key_file>
<ipv6_enable>1</ipv6_enable>
<need_auth>0</need_auth>
<log_enable>1</log_enable>
<log_level>1</log_level>

<http_notify>
<on_connect></on_connect>
<on_play></on_play>
<on_publish></on_publish>
<on_done></on_done>
<notify_method></notify_method>
</http_notify>
<user>
<username>admin</username>
<password>admin</password>
</user>

<output>
<url></url>
<video>
<codec>H264</codec>
<width></width>
<height></height>
<framerate></framerate>
<bitrate></bitrate>
</video>
<audio>
<codec>G711U</codec>
<samplerate></samplerate>
<channels></channels>
<bitrate></bitrate>
</audio>
</output>

<proxy>
<suffix>proxy</suffix>
<url></url>
<user></user>
<pass></pass>
<transfer>TCP</transfer>
<ondemand>0</ondemand>
<output>
<video>
<codec>H264</codec>
<width></width>
<height></height>
 <framerate></framerate>
<bitrate></bitrate>
</video>
<audio>
<codec>AAC</codec>
<samplerate></samplerate>
<channels></channels>
<bitrate></bitrate>
</audio>
</output>
</proxy>

<pusher>
<suffix>pusher</suffix>
<video>
<codec>H264</codec>
</video>
<audio>
<codec>G711U</codec>
<samplerate>8000</samplerate>
<channels>1</channels>
</audio>
<transfer>
<mode>RTSP</mode>
<ip></ip>
<vport>50001</vport>
<aport>50002</aport>
</transfer>
<output>
<video>
<codec></codec>
<width></width>
<height></height>
<framerate></framerate>
 <bitrate></bitrate>
</video>
<audio>
<codec></codec>
<samplerate></samplerate>
<channels></channels>
<bitrate></bitrate>
</audio>
</output>
</pusher>

<backchannel>
<codec>G711U</codec>
<samplerate>8000</samplerate>
<channels>1</channels>
</backchannel>
</config>

4.2 Configuring Node Description

4.2.1 System parameters

<serverip>
Specify the IP address (Support IPv4 and IPv6) or domain name of the RTSP server,
if not specified, the rtsp server will listen to all network interfaces.

<serverport>
Specify the RTSP server port, the default is 554.
Note: On Linux systems, ports below 1024 are reserved by the system and require
root privileges to be used.

<loop_nums>
When streaming media files, specify the number of loop playback,-1 means
infinite loop.
 <multicast>
Whether to enable rtp multicast function, 0-disable, 1-enable.

<udp_base_port>
UDP media transmission base port, RTSP over UDP mode assign UDP port on this
base port.
Each rtsp session needs to assign 8 UDP ports, video RTP/RTCP port, audio
RTP/RTCP port, METADATA stream RTP/RTCP port, audio back-channel RTP/RTCP port.

<metadata>
Whether to enable the meta data stream, 0-disable, 1-enable.

<rtsp_over_http>
Whether to enable rtsp over http function, 0-disable,1-enable.

<http_port>
Specify the HTTP service port for rtsp over http function.
Note: On Linux systems, ports below 1024 are reserved by the system and require
root privileges to be used.

<rtsp_over_https>
Whether to enable rtsp over https function, 0-disable,1-enable.

<https_port>
Specify the HTTPS service port for rtsp over https function.
Note: On Linux systems, ports below 1024 are reserved by the system and require
root privileges to be used.

<cert_file>
Specify the HTTPS service certificate file.

<key_file>
Specify the HTTPS service key file.
 Note: The certificate file ssl.ca and key file ssl.key provided by default are
self signed local hosts certificates, only for testing purposes (browsers may pop
up untrusted certificate warnings), and cannot be used in formal deployment
environments.

<ipv6_enable>
Indicates whether IPv6 is enabled, 0-disable, 1-enable.
Note: If the server does not specify a server ip in <serverip> and the
<ipv6_enable> is 1, and the server has an IPv6 address, the client can connect to
the server through the IPv6 address.

<need_auth>
Whether enable the user authentication function,0-disable,1-enable.

<log_enable>
Whether enable the log function,0-disable,1-enable.

<log_level>
The log level:
TRACE 0
DEBUG 1
INFO 2
WARN 3
ERROR 4
FATAL 5

4.2.2 http_notify
<http_notify> : Specify the HTTP notification callback address.
<on_connect> :
Sets HTTP connection callback. When clients issues connect command an
HTTP request is issued and command processing is suspended until it returns
result code. If HTTP 200 code is returned then RTSP session continues.
HTTP request receives a number of arguments. POST method is used with
application/x-www-form-urlencoded MIME type. The following arguments are
 passed to caller:
protocol=rtsp
call=connect
addr - client IP address
url - URL requested by the client
name – stream name
clientid – rtsp session id
<on_play> :
Sets HTTP play callback. Each time a clients issues play command an HTTP
request is issued and command processing is suspended until it returns
result code. If HTTP 200 code is returned then RTSP session continues.
HTTP request receives a number of arguments. POST method is used with
application/x-www-form-urlencoded MIME type. The following arguments are
passed to caller:
protocol=rtsp
call=play
addr - client IP address
url - URL requested by the client
name – stream name
clientid – rtsp session id
<on_publish> :
The same as on_play above with the only difference that this node sets
callback on publish command. Instead of remote pull push is performed in
this case.
<on_done> :
Sets play/publish terminate callback. All the above applies here.
However HTTP status code is not checked for this callback.
<notify_method>
Sets HTTP method for notifications. Default is POST
with application/x-www-form-urlencoded content type.
Support GET and POST method.

4.2.3 User node

<user> : Specify the login username password, it can configure multiple nodes
 <username>
The login username.

<password>
The login password.

4.2.4 Output node

<output> : Specify the audio and video output parameters, it can configure
multiple nodes.
<url>
Match URL address, it can be filename, or file extension name, or special
suffix. Such as:
screenlive : match living screen stream
videodevice : match camera video stream
*.mp4 : match all mp4 media file
sample.flv : match sample.flv file
If not config this node, it will match all url as the audio/video default
output parameters.
The match order from top to bottom, therefore the default output
configuration should be placed in the last.

<video> : Specify the video output parameters

<codec>
Specify the video stream codec, it can specify the following value:
H264 : output H264 video stream
H265 : output H265 video stream
MP4: output MP4 video stream
JPEG: output MJPEG video stream

<width>
Specify the output video width, If 0 use the original video width
(live screen stream use the screen width, camera stream use the default
width)
 <height>
Specify the output video height, If 0 use the original video height
(live screen stream use the screen height, camera stream use the default
height)
<framerate>
Specify the output video framerate, If 0 use the original video
framerate (live screen use the default value 25, camera stream use the
default value 25)
<bitrate>
Specify the output video bit rate, if 0, automatically calculate
the output bit rate, the unit is kb/s.
Note: This parameter is valid only if encoding is required (eg
screenlive, videodevice) or transcoding is required.

<audio> : Specify the audio output parameters

<codec>
Specify the audio stream codec, it can specify the following value:
G711A: output G711 a-law audio stream
G711U: output G711 mu-law audio stream
G722: output G722 audio stream
G726: output G726 audio stream
AAC: output AAC audio stream
OPUS: output OPUS audio stream

<samplerate>
Specify the audio sample rate, it can specify the following values:
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
If 0 use the original audio sample rate (audio device stream use
the default value 8000)
<channels>
Specify the audio channel number, 1 is mono, 2 is stereo
If 0 use the original audio channel number (audio device stream use
the default value 2)
Note : G726 only support mono.
 <bitrate>
Specify the output audio bit rate, if 0, automatically calculate
the output bit rate, the unit is kb/s.
Note: This parameter is valid only if encoding is required (eg
screenlive, videodevice) or transcoding is required.

4.2.5 Proxy node

<proxy> : Specify the rtsp proxy parameters, it can configure multiple nodes
<suffix>
Specify the rtsp stream suffix, you can play the proxy stream from:
rtsp://[serverip]:[serverport]/[suffix]
<url>
The original rtsp/rtmp/srt/http mjpeg stream address.
<user> <pass>
Specify the original rtsp/rtmp/srt/http mjpeg stream login user and
password.
<transfer>
Specify the rtsp client transfer protocol:
TCP: rtsp client uses RTP over TCP
UDP: rtsp client uses RTP over UDP
MULTICAST: rtsp client uses multicast
<ondemand>
Connect on demand, 1-Connect when needed, 0-Always keep connected
<output>
Specify the stream output parameter. If the parameter does not appear,
use the parameters of the original RTSP/RTMP/SRT/HTTP MJPEG stream. If it
appears and the configured parameters are inconsistent with the parameters
of the original RTSP/RTMP/SRT/HTTP MJPEG stream, then the transcode output
is performed.
The child nodes under this node are consistent with the meaning of the
<output> node.
4.2.6 Pusher node
<pusher> : Specify the data pusher parameters, it can configure multiple nodes
<suffix>
Specify the rtsp stream suffix, you can play the pusher stream from:
rtsp://[serverip]:[serverport]/[suffix]

<video> : Specify the input video data parameters

<codec>
Specify the video codec, it can specify the following value:
H264 : H264 video stream
H265 : H265 video stream
JPEG: MJPEG video stream
MP4: MPEG4 video stream

<audio> : Specify the input audio data parameters

<codec>
Specify the audio codec, it can specify the following value:
G711A: G711 a-law audio stream
G711U: G711 mu-law audio stream
G722: G722 audio stream
G726: G726 audio stream
OPUS: OPUS audio stream
AAC: AAC audio stream
<samplerate>
Specify the audio sample rate, it can specify the following values:
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
<channels>
Specify the audio channel number, 1 is mono, 2 is stereo
Note : G726 only support mono.

<transfer>：Specify the data transfer parameters

<mode>：Specify the data transer protocol, it can specify the following
value:
TCP: Use TCP connection to transfer the data
 UDP: Use UDP connection to transfer the data
RTSP: Use RTSP connection to transfer the data. It supports standard
rtsp push, such as FFMPEG rtsp push.

<ip>: Specify data receiving IP address, if there is no configuration,

the default IP address is used.
<vport>: Specify the video data receiving port
<aport>: Specify the audio data receiving port
Note : <ip>, <vport>, <aport> these parameters are valid when <mode>
is TCP or UDP.

<output>
Specify the stream output parameter. If the parameter does not appear,
use the parameters of the original pusher stream. If it appears and the
configured parameters are inconsistent with the parameters of the original
pusher stream, then the transcode output is performed.
The child nodes under this node are consistent with the meaning of the
<output> node.

4.2.7 Backchannel node

<backchannel> : specify the audio back channel parameters
<codec>
Specify the audio back channel stream codec, it can specify the
following value:
G711A: G711 a-law audio stream
G711U: G711 mu-law audio stream
G722: G726 audio stream
G726: G726 audio stream
OPUS: OPUS audio stream
AAC: AAC audio stream

<samplerate>
Specify the audio back channel sample rate, it can specify the following
values:
 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
If 0 use the default value 8000

<channels>
Specify the audio channel number, 1 is mono, 2 is stereo
If 0 use the default value 1
Note : G726 only support mono.
 Chapter 5 Data pusher
Data push means that RTSP server receives external data sources and then sends
them out as RTSP streams.
The data pusher supports TCP, UDP and RTSP mode.
Audio and video data are packaged and sent in RTP format.
If it is TCP mode, you need to add 4 bytes in front of the RTP header, as the
following:

typedef struct
{
uint32 magic : 8;
uint32 channel : 8;
uint32 rtp_len : 16;
} RILF;

magic: 0x24
channel: 0
rtp_len: the RTP payload length, including RTP header

Note: If you use TCP or UDP mode data push, you need to add <pusher> tag in the
rtsp server configuration file, specify the push audio and video parameters and push
port, etc.
If you use RTSP mode to push data, no configuration is required. The url suffix
of the pushed RTSP address can be any legal string.

If it is RTSP mode, it supports standard RTSP push stream, such as FFMPEG rtsp
push.
FFMPEG rtsp over UDP:
ffmpeg -re -i test.mp4 -vcodec libx264 -acodec copy -preset ultrafast -f rtsp
rtsp://[serverip]:[serverport]/pusher

FFMPEG rtsp over TCP:

ffmpeg -re -i test.mp4 -vcodec libx264 -acodec copy -preset ultrafast -f rtsp -rtsp_transport
tcp rtsp://[serverip]:[serverport]/pusher
 After the above ffmpeg rtsp push command is running, you can use the following
address to play the rtsp stream:
rtsp://[serverip]:[serverport]/pusher
 Chapter 6 RTSP over HTTP
The key of RTSP over HTTP is to allow RTSP packets to communicate via HTTP port.
We know that the standard port of RTSP is 554, but due to various security policy
configurations such as firewalls, there may be restrictions when the client accesses
port 554, which prevents the normal transmission of RTSP packets.
But the HTTP port (port 80) is generally open, so there is the idea of letting
RTSP packets pass through port 80, namely RTSP over HTTP.
The details of RTSP over HTTP are as follows:
First, the client opens two socket connect to the rtsp server HTTP ports. We
call these two sockets "data socket" and "command socket".
Step 1. The client sends an HTTP GET command through the "data socket" to request
an RTSP connection.
Step 2. The server responds to the HTTP GET command through the "data socket"
and responds with success/failure.
Step 3. The client creates a "command socket" and sends an HTTP POST command
through the "command socket" to establish an RTSP session.
At this point, the auxiliary function of HTTP is completed, and the server does
not return the client's HTTP POST command. Next is the standard process of RTSP on
the HTTP port, but it needs to be completed through two sockets. The "command socket"
is only responsible for sending, and the "data socket" is only responsible for
receiving.
Step 4. The client sends RTSP commands (BASE64 encoding) through the "command
socket".
Step 5. The server responds to the RTSP command (in plain text) through the "data
socket".
Step 6. Repeat Step4-Step5 until the client sends the RTSP PLAY command and the
server responds to the RTSP PLAY command.
Step 7. The server transmits audio and video data to the client through the "data
socket"
After the data exchange is complete...
Step 8. The client sends the RTSP TEARDOWN command (BASE64 encoding and) through
the "command socket"
Step 9. The server responds to the RTSP TEARDOWN command (in plain text) through
the "data socket".
Step 10. Close the two sockets.
VLC supports RTSP over HTTP, the settings as the follows:

Happytime rtsp client

(http://happytimesoft.com/products/rtsp-client/index.html) supports RTSP
over HTTP, The setting as the following:

Happytime rtsp client also supports rtsp streams starting with http:// or
https://.
If it starts with http://, it is considered to be a rtsp over http stream.
If it starts with https://, it is considered to be a rtsp over https stream.
 Chapter 7 RTSP over Websocket
First establish an HTTP connection, and then upgrade to the websocket protocol，
RTSP over websocket protocol upgrade process:
C-->S:
GET /websocket HTTP/1.1 Host: 192.168.3.27
Upgrade: websocket Connection: Upgrade
Sec-WebSocket-Key: KSO+hOFs1q5SkEnx8bvp6w== Origin: http://192.168.3.27
Sec-WebSocket-Protocol: rtsp.onvif.org Sec-WebSocket-Version: 13
S-->C:
HTTP/1.1 101 Switching Protocols Upgrade: websocket
Connection: Upgrade
Sec-WebSocket-Accept: G/cEt4HtsYEnP0MnSVkKRk459gM= Sec-WebSocket-Protocol:
rtsp.onvif.org
Sec-WebSocket-Version: 13
After the protocol upgrade is successful, perform normal rtsp protocol exchange,
and send and receive data through websocket connection.
Happytime rtsp client
(http://happytimesoft.com/products/rtsp-client/index.html) supports RTSP
over Websocket, The setting as the following:

Happytime rtsp client also supports rtsp streams starting with ws://.
If it starts with ws://, it is considered to be a rtsp over websocket stream.
 Chapter 8 RTP Multicast
To enable the rtp multicast function, it need to specify the <multicast> to 1
in the configuration file.
The rtsp server does not support the configuration of multicast addresses.
Different rtsp stream addresses use multicast, randomly assigned multicast
addresses starting with 232.
Different rtsp sessions use rtp multicast to play the same rtsp stream, using
the same multicast address.Only the first rtsp session sends audio and video data,
and subsequent sessions refer to the first rtsp session.
The rtp multicast stream address is the same as the other rtsp stream address.
Use VLC to test rtp multicast, use the following settings:
 Chapter 9 Audio back channel
The backchannel connection handling is done using RTSP [RFC 2326]. Therefore
a mechanism is introduced which indicates that a client wants to built up a
backchannel connection. RTSP provides feature-tags to deal with such functionality
additions. A device that supports bi-directional connections (e.g audio or metadata
connections) shall support the introduced RTSP extensions.

9.1 RTSP Require- Tag

The RTSP standard [RFC 2326] can be extended by using additional headers objects.
For that purpose a Require tag is introduced to handle special functionality
additions (see [RFC 2326], 1.5 Extending Rtsp and 12.32 Require).
The Require-tag is used to determine the support of this feature. This header
shall be included in any request where the server is required to understand that
feature to correctly perform the request.
A device that supports backchannel and signals Audio output support via the
AudioOutputs capability shall understand the backchannel tag:
www.onvif.org/ver20/backchannel
An RTSP client that wants to built up an RTSP connection with a data backchannel
shall include the Require header in its requests.

9.2 Connection setup for a bi- directional connection

A client shall include the feature tag in it's DESCRIBE request to indicate that
a bidirectional data connection shall be established.
A server that understands this Require tag shall include an additional media
stream in its SDP file as configured in its Media Profile.
An RTSP server that does not understand the backchannel feature tag or does not
support bidirectional data connections shall respond with an error code 551 Option
not supported according to the RTSP standard. The client can then try to establish
an RTSP connection without backchannel.
A SDP file is used to describe the session. To indicated the direction of the
media data the server shall include the a=sendonly in each media section representing
media being sent from the client to the server and a=recvonly attributes in each
media section representing media being sent from the server to the client.
The server shall list all supported decoding codecs as own media section and
the client chooses which one is used. The payload type and the encoded bitstream
shall be matched with one of the a=rtpmap fields provided by the server so that the
server can properly determine the audio decoder.

This SDP file completely describes the RTSP session. The Server gives the client
its control URLs to setup the streams.
In the next step the client can setup the sessions:
 The third setup request establishes the audio backchannel connection.
In the next step the client starts the session by sending a PLAY request.

After receiving the OK response to the PLAY request the client MAY start sending
audio data to the server. It shall not start sending data to the server before it
has received the response.
The Require-header indicates that a special interpretation of the PLAY command
is necessary. The command covers both starting of the video and audio stream from
NVT to the client and starting the audio connection from client to server.
To terminate the session the client sends a TEARDOWN request.
9.3 Example
Server with Onvif backchannel support (with multiple decoding capability)
If a device supports multiple audio decoders as backchannel, it can signal such
capability by listing multiple a=rtpmap fields illustrated as follows.
 Chapter 10 Run RTSP Server
The server is a console application.
Windows: to run the server, simply type "rtspserver".
Linux: to run the server, type "./start.sh", on linux platform, rtsp server run
as deamon by default.

rtsp server supports the following command line options:

-c config specify the configuration file
-c option specifies the configuration file,if not specified, the default
configuration rtspserver.cfg is used.
-l [device|videodevice|audiodevice|window]
-l device list available video and audio capture device
-l videodevice list available video capture device
-l audiodevice list available audio capture device
-l window list available application window

Below is sample output of -l device:

rtspserver -l device

Available video capture device :

index : 0, name : FaceTime HD Camera (Built-in)

Available audio capture device :

index : 0, name : Headset Microphone (Apple Audio Device)
index : 1, name : Internal Digital Microphone (Apple Audio Device)

Note : The demo version has the following limitations:

Maximum support four concurrent sessions.
 Chapter 11 Multiple capture devices support
If your system have multiple audio capture device, you can use
rtsp://[serverip]:[serverport]/audiodeviceN
The N to specify the audio capture device index, start from 0, such as:
rtsp://192.168.0.100/audiodevice ; stream audio from the first audio device
rtsp://192.168.0.100/audiodevice1 ; stream audio from the second audio device

If your system have multiple video capture device, you can use
rtsp://[serverip]:[serverport]/videodeviceN
The N to specify the video capture device index, start from 0, such as:
rtsp://192.168.0.100/videodevice ; stream video from the first video device
rtsp://192.168.0.100/videodevice1 ; stream video from the second video device

If your system have multiple monitors, you can use

rtsp://[serverip]:[serverport]/screenliveN
The N to specify the monitor index, start from 0, such as:
rtsp://192.168.0.100/screenlive ; stream living screen from the first monitor
rtsp://192.168.0.100/screenlive1 ; stream living screen from the second monitor

The audio index or video index represents which device can run the following
command to view:
rtspserver -l device

videodevice or audiodevice can also specify the device name, such as :

rtsp://[serverip]:[serverport]/videodevice=testvideo

Run the rtspserver -l device command to get the device name.

Note that there can be no spaces in the device name, if the device name contains
spaces, you need to use %20 instead of spaces.

If the device name is “FaceTime HD Camera”, the rtsp stream address is:
rtsp://[serverip]:[serverport]/videodevice=FaceTime%20HD%20Camera
 Chapter 12 Capture application window
The rtsp server supports capturing application windows, you can use the
following command to list valid application windows:
rtspserver -l window

Below is a sample output of the command

Available window name :

C:\Windows\system32\cmd.exe - RtspServer.exe -l window

user manual.doc - WPS Office
Rtsp-server Project - Source Insight - [Main.cpp]
RtspServer

You can use the following url to capture the specified application window:
rtsp://[serverip]:[serverport]/window=[window title]
Note : window title case insensitive

Such as :
rtsp://[serverip]:[serverport]/window=rtspserver

Note that there can be no spaces in the window title, if the window title contains
spaces, you need to use %20 instead of spaces. Such as:
rtsp://[serverip]:[serverport]/window=user%20manual.doc%20-%20WPS%20Office
 Chapter 13 Support URL parameters
RTSP server supports URL parameters, with the following format:
Taking playing the test.mp4 file as an example:
rtsp://[serverip]:[serverport]/test.mp4?param1=value1&param2=value2
Pararm1 and param2 represent URL parameters, while value1 and value2 represent
the values of param1 and param2.
Note : The parameter value specified through the URL has a higher priority than
the parameters configured in the configuration file.

The RTSP server supports the following parameters:

srtp : Whether to enable SRTP

Possible values
0 - disable srtp
1 - enable srtp

t : Specify the transmission method

Possible values:
unicast
multicast

p ： Specify transmission protocol

Possible values:
udp : rtp over udp
tcp : rtp over tcp

ve : Specify video encoding

Possible values:
H264
H265
JPEG
MP4
 fps ：Specify video frame rate
w : Specify video width
h : Specify video height
vb ：Specify video bitrate

ae ：Specify audio encoding

Possible values:
PCMU ： g711 ulaw
PCMA: g711 alaw
G726
G722
OPUS
AAC

sr : Specify audio samplerate

ch ：Specify the number of audio channels
ab ：Specify audio bitrate

Example:
Using unicast RTP over UDP mode, output video encoding as H264, resolution as
1920 * 1080, frame rate as 30, bit rate as 4000K, audio as AAC, sampling rate as
44100, stereo RTSP stream:
rtsp://[serverip]:[serverport]/test.mp4?t=unicast&p=udp&ve=h264&w=1920&h=1
080&fps=30&vb=4000&ae=aac&sr=44100&ch=2
 Chapter 14 Support SRTP
Happytiem rtsp server supports Secure Real-time Transport Protocol (secure RTP
or SRTP) for publishing and playback. The encryption keys can be passed in the SDP
data. The open-source tools FFmpeg and FFplay can be used for SRTP testing.

14.1 SRTP for publishing

Happytime RTSP Pusher supports SRTP publishing. Modify the RTSP Pusher
configuration file to set <SRTP> under the <Pusher> tag to 1, and RTSP Pusher will
use SRTP for data publishing:
RTSP Pusher uses SRTP to publish configuration examples:

<pusher>
<src>test.mp4</src>
<transfer>
<mode>RTSP</mode>
<rtspurl>rtsp://192.168.3.36/myapp/live</rtspurl>
<user>admin</user>
<pass>admin</pass>
</transfer>
<video>
……
</video>
<audio>
……
</audio>
<metadata>0</metadata>
<srtp>1</srtp>
</pusher>

You will see SDP information in the ANNOUNCE request message of rsp pusher. It
should look something like this:

v=0
o=- 0 0 IN IP4 192.168.3.36
c=IN IP4 192.168.3.36
s=session
t=0 0
 a=control:*
m=video 0 RTP/SAVP 96
a=rtpmap:96 H264/90000
a=fmtp:96
packetization-mode=1;profile-level-id=42801F;sprop-parameter-sets=Z0KAH5ZSAKALdJQEBAUAAAMAAQ
AAAwAyhA==,aMuNSA==
a=control:realvideo
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:KSO+hOFs1q5SkEnx8bvp67Om2zyHDD6ZJF4NHAa3
m=audio 0 RTP/SAVP 97
a=rtpmap:97 MPEG4-GENERIC/44100/2
a=fmtp:97
streamtype=5;profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;
config=121056E500
a=control:realaudio
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R96zEk3IQ7uLph8DWn0JOCUfXdTL/Jb1RTsTDYkK

14.2 SRTP for playback

The RTSP server supports the URL parameter SRTP to specify whether SRTP playback
is enabled or not.
Example:
Enable SRTP to play test.mp4 files:
rtsp://[serverip]:[serverport]/test.mp4?srtp=1
Use FFplay or Happytime rtsp client to test playback.