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From other server via RTMP

Overview

WCS can capture on demand an RTMP video stream published by another server. The captured stream can be played on any of supported platforms using any of supported technologies. RTMP stream capturing is managed using REST API.

Operation flowchart

  1. The /pull/rtmp/pull REST query is sent to the WCS server
  2. The WCS server requests the RTMP stream from the specified server
  3. The RTMP stream is broadcast to the WCS server
  4. The browser requests playing the captured stream via WebSocket
  5. The browser receives the stream via WebRTC

REST API

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

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

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
  • /pull/rtmp/pull is the REST method used

REST queries and responses

REST method Request body Response body Response status Description
`/pull/rtmp/pull` 
{ 
    "uri":"rtmp://myserver.com/live/myStream",
    "localStreamName":"stream1",
    "record": "false",
    "hasAudio": true,
    "hasVideo": true
}
409 Conflict 500 Internal error Pull the RTMP stream at the specified URL
`/pull/rtmp/find_all` 
[
    {
        "localMediaSessionId": "5a072377-73c1-4caf-abd3"
        "localStreamName": "stream1"
        "uri": "rtmp://myserver.com/live/myStream",
        "status": "PROCESSED_REMOTE",
        "hasAudio": true,
        "hasVideo": true,
        "record": false
    }
]
200 OK 404 Not found 500 Internal error Find all pulled RTMP streams
`/pull/rtmp/terminate` 
{ 
    "uri":"rtmp://myserver.com/live/myStream"
}
200 OK 404 Not found 500 Internal error Terminate the pulled RTMP stream

Parameters

Parameter Description Example
uri URL of the RTMP stream `rtmp://myserver.com/live/myStream`
localMediaSessionId Session identifier `5a072377-73c1-4caf-abd3`
localStreamName Local name assigned to the captured stream. By this name the stream can be requested from the WCS server `stream1`
status Current stream status `PROCESSED_REMOTE`
record Is the pulled stream recording `false`
hasAudio Stream has audio track `true`
hasVideo Stream has video track `true`

Pulled stream publishing with a given name

Since build 5.2.724 it is possible to set the name to publish stream on server using localStreamName query parameter. If the parameter is not set, the stream name will be set to uri, as done in previous builds.

RTMP stream repeatedly capturing with the same URI

/pull/rtmp/pull query returns 409 Conflict when trying to repeatedly capture RTMP stream with the same URI. If the stream is already published on the server, it is necessary to subscribe to it.

Configuration

Codecs parameters definition

The SDP description file for the RTMP agent rtmp_agent.sdp placed to /usr/local/FlashphonerWebCallServer/conf folder may be used to change audio codec parameters:

o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
m=video 0 RTP/AVP 95
a=rtpmap:95 H264/90000
a=fmtp:95 profile-level-id=42e01f;packetization-mode=1
a=sendonly
m=audio 0 RTP/AVP 103 96 97 98 99 100 102 108 104
a=rtpmap:108 mpeg4-generic/48000/1
a=rtpmap:96 mpeg4-generic/8000/1
a=rtpmap:97 mpeg4-generic/11025/1
a=rtpmap:98 mpeg4-generic/12000/1
a=rtpmap:99 mpeg4-generic/16000/1
a=rtpmap:100 mpeg4-generic/22050/1
a=rtpmap:104 mpeg4-generic/24000/1
a=rtpmap:102 mpeg4-generic/32000/1
a=rtpmap:103 mpeg4-generic/44100/1
a=recvonly

To enable recording of both audio and video (instead of audio only) during captured stream recording specify the following attribute in this file

a=sendonly

for video.

Publishing a stream without audio or video

If audio or video description is removed from rtmp_agent.sdp, an RTMP stream captured will be published on WCS without audio or video respectively. For example, use the following SDP to publish video only:

v=0
o=- 1988962254 1988962254 IN IP4 0.0.0.0
c=IN IP4 0.0.0.0
t=0 0
a=sdplang:en
m=video 0 RTP/AVP 95
a=rtpmap:95 H264/90000
a=fmtp:95 profile-level-id=42e01f;packetization-mode=1
a=sendonly

Frame type detection

Sometimes, RTMP stream source may set a media frame type incorrectly at RTMP protocol level. In this case frame type should be detected according to frame content. To enable this, use the following parameter added in build 5.2.1446

rtmp_detect_h264_frame_type=true

Quick manual on testing

Capturing of an RTMP stream broadcast by another server using the REST-query /pull/rtmp/pull

  1. For the test we use:
  2. the demo server at demo.flashphoner.com;
  3. the Chrome browser and the REST-client to send queries to the server;

  4. the Two Way Streaming web application to play the captured stream in a browser.

  5. Open the REST client. Send the /pull/rtmp/pull query and specify the URL of the RTMP stream in parameters:

  6. Make sure the stream is captured by the server. To do this, send the /rtmp/pull/find_all request:

    and copy the local name of the stream from the localStreamName response parameter:

  7. Open the page of the Two Way Streaming web application. Click Connect and specify the local stream name, then click Play:

  8. WebRTC internals diagrams in a browser:

Capturing of an RTMP stream broadcast by another server without using REST API

  1. For the test we use:
  2. the demo server at demo.flashphoner.com;

  3. the web application, Two Way Streaming, to capture and play the captured stream in a browser.

  4. Open the page of the Two Way Streaming web application. Click Connect and specify the name of the RTMP stream you want to capture, then click Play:

  5. WebRTC internals diagrams in a browser:

Call flow

Below is the call flow when capturing an RTMP stream from another server

Authentication on a source server

WCS supports Adobe authentication on RTMP server while capturing a stream from it using RTMP URL parameters:

rtmp://username:password@server:1935/live/streamKey

Note that if an RTMP stream is requested from another WCS server, this kind of authentication is not supported.

Since build 5.2.1069 it is possible to pass authentication parameters after stream name

rtmp://server:1935/live/streamKey?user=username&password=password

In this case the parameters will be passed to RTMP server in connect message.

The parameters can be set after application name too

rtmp://server:1935/live?user=username&password=password/streamKey

In this case the parameters will also be passed to RTMP server in connect message. If the stream is requested from another WCS server via RTMP, authentication parameters will be available in REST hook /connect.

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

Symptoms

There are warnings 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

Enable Fraunhofer AAC codec with the following parameter

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 recording such 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. Some RTMP functions does not supported and will be ignored

  • FCSubscribe
  • FCPublish
  • FCUnpublish
  • onStatus
  • onUpstreamBase
  • releaseStream

6. When recording the captured stream to the disk, only audio is recorded

Symptoms

When the "record": true parameter is set in the /pull/rtmp/pull REST query, the received file lacks video track, only audio is present.

Solution

In the SDP settings set the foolowing attribute

a=sendonly

for the video track.

7. Freezes are possible when RTMP stream is captured from Adobe Media Server with Aggregate messages enabled

Symptoms

Freezes may occur, publishing may fail by timeout when capturing RTMP stream from Adobe Media Server

Solution

Disable Aggregate messages in AMS configuration by switching off the following parameters:

  • all occurences of EnableAggMsgs in Server.xml
  • Client/AggregateMessages and Queue/AggregateMessages in conf/_defaultRoot_/_defaultVHost/Application.xml
  • AggregateMessages in Vhost.xml