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Overview

A video stream is captured from an RTSP source that provides audio and video in the supported codecs. Then, the server transforms this video stream for playing in browsers and mobile devices.

RTSP sources

  • IP cameras
  • Media servers
  • Surveillance systems
  • Conference servers

Supported codecs

  • H.264
  • VP8
  • AAC
  • G.711
  • Speex

Supported platforms and browsers


Chrome

Firefox

Safari 11

Internet Explorer

Edge

Windows

+

+


+

+

Mac OS

+

+

+



Android

+

+




iOS

-

-

+



Operation flowchart


1. The browser establishes a connection to the server via the Websocket protocol and sends the play command.

2. The server connects to the RTSP source and send the play command.

3. The RTSP source sends the RTSP stream to the server.

4. The server transforms the stream to WebRTC and gives the stream to the browser.

Configuration

RTSP client binding to certain address

Sometimes, when IP camera should be connected through VPN, RTSP client should be bound to certain IP address. The option rtsp_client_address in settings file flashphoner.properties defines this address, for example:

rtsp_client_address=172.16.0.3

RTSP stream capturing via UDP

By default, RTSP streams are captured via TCP. Stream capturing over UDP can be turned on if necessary with the following parameter

rtsp_interleaved_mode=false

Audio and video track selection in RTSP stream

By default, audio and video tracks in RTSP tream are selected dynamically according to camera SDP. Tracks order can be set explicitly if necessary with the following parameter, for example

rtsp_interleaved_channels=2-3;0-1

Where

  • 2-3 - audio track channels
  • 0-1 - video track channels

Playing RTSP stream in AnnexB form

Some IP cameras are publishing H264 stream in AnnexB format, Honeywell MAXPRO Video Streamer for example. To play video from such cameras, the following parameter is added since build 5.2.636 to enable AnnexB format parsing

h264_check_and_skip_annexb=true

Since build 5.2.946, this parameter is removed from settings, and AnnexB frames are detected and played automatically.

Audio codecs exclusion

Sometimes it is necessary to capture a stream from camera without audio, or disable some audio codecs to escapr sound transcoding. This can be done with the following parameter listing audio codec names to be excluded while capturing stream, for example

rtsp_client_strip_audio_codecs=PCMA,PCMU

This setting will exclude PCMA (alaw) and PCMU (ulaw) codecs. A stream with audio track in those codecs will be captured from a camera as video only.

The listed codecs are excluded at SDP level, by names.

H264 packetization mode configuration

According to H264 specifications, if H264 packetization mode is not set explicitly in stream SDP, if should be set to 0. However, some RTSP cameras may send a stream encided with packetization mode 1, and may omit this mode setting in SDP. This behaviour may lead to stream transcoding and picture quality dropping while playing the stream in Safari browser.

Since build 5.2.820 it is possible to set default packetization mode to use such cameras with the following parameter

default_packetization_mode=1

Quick manual on testing

Capturing of a video stream from the IP camera and playing it in a browser

1. For this test we use:

  • the demo server at demo.flashphoner.com;
  • the Player web application to play the captured stream in the browser.

2. Open the Player web app and specify the URL of the camera in the "Stream" field:


3. Click the "Start" button. Broadcasting of the captured stream begins.


4. WebRTC internals diagrams:

Stream capture from the IP camera management by REST API

Usually, it is enough to set the camera URL as stream name to capture stream from IP camera. However, it is possible to manage RTSP stream capture by REST API if necessary.

Testing

1. For this test we use:

2. Open the REST client. Send the /rtsp/startup query specifying the URL of the web camera in parameters:


3. Make sure the stream is captured by the server. To do this, send the /rtsp/find_all query:


4. Open the Player web app and in the "Stream" field specify the URL of the web camera and click Start. Browser starts to play the stream:


5. Send the /rtsp/terminate query specifying the URL of the web camera in parameters:


6. Stream playback will terminate displaying an error:

REST-queries

A REST-query should be HTTP/HTTPS POST request as follows:

  • HTTP: http://test.flashphoner.com:8081/rest-api/rtsp/startup
  • HTTPS: https://test.flashphoner.com:8444/rest-api/rtsp/startup

Where:

  • test.flashphoner.com - is the address of the WCS server
  • 8081 - is the standard REST / HTTP port of the WCS server
  • 8444 - is the standard HTTPS port
  • rest-api - is the required part of the URL
  • /rtsp/startup - REST-method to use

REST-methods and response statuses

REST-method

Example of REST-query

Example of response

Response statuses

Description

/rtsp/startup

{
 "uri":"rtsp://myserver.com/live/myStream",
 "localStreamName": "myRTSPstream"
}

409 - Conflict

500 - Internal error


Pull the RTSP stream by the specified URL


/rtsp/find_all


{
"uri": "rtsp://myserver.com/live/myStream",
"status": "PLAYING",
"localStreamName": "myRTSPstream"
}

200 – streams found
404 – streams not found


Find all pulled RTSP-streams

/rtsp/terminate

{
"uri":"rtsp://myserver.com/live/myStream"
}

200 - stream terminated
404 - stream not found


Terminate the pulled RTSP stream

Parameters

Parameter name

Description

Example

uri

URL of the RTSP stream

rtsp://myserver.com/live/myStream

localStreamNameName to set to the stream capturedmyRTSPstream

status

Current status of the stream

PLAYING

RTSP stream repeatedly capturing with the same URI

/rtsp/startup query returns 409 Conflict while trying to repeatedly capture RTSP stream with the same URI. If the stream from the IP camera is already published on the server, it is necessary to subscribe to it.

Call flow

Below is the call flow when using the Player example

player.html

player.js

1. Establishing a connection to the server.

Flashphoner.createSession(); code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function(session){
        setStatus(session.status());
        //session connected, start playback
        playStream(session);
    }).on(SESSION_STATUS.DISCONNECTED, function(){
        setStatus(SESSION_STATUS.DISCONNECTED);
        onStopped();
    }).on(SESSION_STATUS.FAILED, function(){
        setStatus(SESSION_STATUS.FAILED);
        onStopped();
    });

2. Receiving from the server an event confirming successful connection.

ConnectionStatusEvent ESTABLISHED code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function(session){
        setStatus(session.status());
        //session connected, start playback
        playStream(session);
    }).on(SESSION_STATUS.DISCONNECTED, function(){
        ...
    }).on(SESSION_STATUS.FAILED, function(){
        ...
    });

3. Request to play the stream.

session.createStream(), stream.play(); code

IP camera URL is passed to createStream() method as stream name

function playStream(session) {
    var streamName = $('#streamName').val();
    var options = {
        name: streamName,
        display: remoteVideo,
        flashShowFullScreenButton: true
    };
    ...    
    stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
        ...
    });
    stream.play();
}

4. Request from WCS to the RTSP source to broadcast the stream.

5. Broadcasting the RTSP stream

6. Receiving from the server an event confirming successful capturing and playing of the stream.

StreamStatusEvent, статус PLAYING code

    stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
        ...
    }).on(STREAM_STATUS.PLAYING, function(stream) {
        $("#preloader").show();
        setStatus(stream.status());
        onStarted(stream);
        ...
    });
    stream.play();

7. Sending audio- and video stream via WebRTC

8. Stopping playing the stream.

stream.stop(); code

function onStarted(stream) {
    $("#playBtn").text("Stop").off('click').click(function(){
        $(this).prop('disabled', true);
        stream.stop();
    }).prop('disabled', false);
    $("#fullScreenBtn").off('click').click(function(){
       stream.fullScreen();
    }).prop('disabled', false);
    $("#volumeControl").slider("enable");
    stream.setVolume(currentVolumeValue);
}

9. Receiving from the server an event confirming successful stop of the stream playback.

StreamStatusEvent, статус STOPPED code

    stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
        ...
    }).on(STREAM_STATUS.PLAYING, function(stream) {
        ...
    }).on(STREAM_STATUS.STOPPED, function() {
        setStatus(STREAM_STATUS.STOPPED);
        onStopped();
    }).on(STREAM_STATUS.FAILED, function(stream) {
        ...
    }).on(STREAM_STATUS.NOT_ENOUGH_BANDWIDTH, function(stream){
        ...
    });
    stream.play();

RTSP connection reuse

If other subscribers request the stream captured from RTSP IP camera, the previous RTSP connection will be used if all subscribers set the same camera URL. For example, two requests to the same IP camera

rtsp://host:554/live.sdp

and

rtsp://host:554/live.sdp?p=1

are differ, then two RTSP connections will be created if streams from both URLs are requested.

Stream capture authentication

WCS supports RTSP stream capture authentication by user name and password, user data should be set in stream URL, for example

rtsp://user:password@hostname/stream

If name or password contains any special characters, they should be escaped such as

rtsp://user:p%40ssword@hostname/stream

Where

  • user is user name
  • p@ssword is password with character '@', it is escaped in URL.

Another IP address redirection handling

Some IP cameras return 302 Moved Temporarily in response to DESCRIBE or OPTIONS query to redirect a client to another IP address for RTSP connection establishing. WCS supports this feature since build 5.2.179.

In this case, if the IP camera redirects requests to another address, and if client establishes connections separately to this camera and directly to the camera where requests are redirected, it is two different streams for WCS. The pulling agent s are created for every of those streams, and subscribers connect to one of those agents depending on address set on connection establishing.

RTSP pulled stream publishing with a given name

The ability was added to publish RTSP pulled stream with a given name since build 5.2.479. The stream name should be set with toStream parameter of /rtsp/startup REST query, for example

POST /rest-api/rtsp/startup HTTP/1.1
Content-Length: 75
Content-Type: application/json

{
    "toStream": "stream1", 
    "uri": "rtsp://myserver.com/live/myStream"
}

By default, if toStream paremeter is not set, the stream name will be formed from RTSP URI. If RTSP stream with such URI is already pulled, or teh stream with given name is already exists, server returns 409 Conflict.

If the name is set to the RTSP stream pulled, this stream can be played in CDN by its name (by default, CDN playback is not available for RTSP streams because they always are pulled locally).

Known issues

1. A stream containing B-frames does not play or plays with artifacts (latencies, lags)

Symptoms:

  • a stream sent by the RTMP encoder does not play or plays with latencies or lags
  • warnings in the client log:
09:32:31,238 WARN 4BitstreamNormalizer - RTMP-pool-10-thread-5 It is B-frame!

Solution

  • change the encoder settings so, that B-frames were not used (lower encoding profile, specify in the command line etc)
  • transcode the stream, in this case there will be no B-frames in transcoded stream

2. AAC frames of type 0 are not supported by decoder and will be ignored while stream pulled playback

In this case, warnings will be displayed in the client log:

10:13:06,815 WARN AAC - AudioProcessor-c6c22de8-a129-43b2-bf67-1f433a814ba9 Dropping AAC frame that starts with 0, 119056e500

Solution: use Fraunhofer AAC codec with the following parameter in flashphoner.properties file

use_fdk_aac=true

3. When publishing and then playing and recording H264 + AAC stream video may be out of sync with sound, or no sound at all.

Symptoms: when playing H264 + AAC stream published on server, and when recordingsuch stream, sound is out of sync with video or absent

Solution:

a) set the following parameter in flashphoner.properties file

disable_drop_aac_frame=true

This parameter also turns off AAC frames dropping.

b) use Fraunhofer AAC codec

use_fdk_aac=true

4. Sound may be distorted or absent when resampled to 11025 Hz

Symptoms: when H264 + AAC stream published on WCS server is played with AAC sample rate 11025 Hz, sound is distorted or absent

Solution: do not use 11025 Hz sample rate, or escape AAC sound resampling to this rate, for example, do not set this sample rate in SDP settings.

5. Connection to the IP camera is lost on error in any track (audio or video)

Symptoms: connection to the IP camera is lost if one of tracks returns error 4**.
Solution: this behavior is enabled by default. However if one-time errors are not critical and should not terminate broadcasting, in the flashphoner.properties files set

rtsp_fail_on_error_track=false
rtp_force_synchronization=true

6. All the characters in a stream name, that are not allowed in URI, should be escaped

Symptoms: RTSP stream is not played with 'Bad URI' error
Solution: any character that is not allowed in URI, should be escaped in stream URL, for example

rtsp://hostname/c@@lstream/channel1

should be set as

rtsp://hostname/c%40%40lstream/channel1

7. Some IP cameras do not support cnonce field in RTSP connection message header.

Symptoms: RTSP stream is played with VLC, but is not played with WCS.

Solution: set the following parameter in flashphoner.properties file

rtsp_auth_cnonce=

with empty value.

8. Streams from some cameras cannot be played due to buffer size lack to write RBSP

Symptoms: RTSP stream is not playing with exceptions in server log

13:10:16,988 ERROR  BitstreamNormalizer - pool-56-thread-1 Failed to normalize SPS 674d002a95a81e0089f950
java.lang.RuntimeException: Failed to write sps rbsp

Solution: increase RBSP buffer size setting (1.5 by default)

h264_sps_rbsp_scale=2

9. Some camera streams can loose audio and video sync

Symptoms: RTSP stream freezes or cannot be played by HLS (some segments cannot be written), in stream statistics there are ubnormally big AV syncronization values

streams_synchronization=camera1/-21800;camera2/2079600704

Solution: increase synchronization buffers for audio and video tracks

audio_incoming_buffer_size=100
video_incoming_buffer_size=100