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REST-method | Example of REST-query | Example of REST response | Response status | Description | |||||||
|---|---|---|---|---|---|---|---|---|---|---|---|
/pull/pull |
| 409 - Conflict 500 - Internal error | Pull the RTMP stream at the specified URL | ||||||||
/pull/find_all |
| 200 – streams are found 500 - Internal error | Find all pulled RTMP streams | ||||||||
/pull/terminate |
| 200 - stream terminated 500 - Internal error | Terminate the pulled RTMP stream |
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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 while 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.
Audio only or video only stream publishing
Since build 5.2.1439 it is possible to set in REST query how to publish an RTMP stream captured on WCS. For example, to publish a video only stream, set
| Code Block | ||||
|---|---|---|---|---|
| ||||
{
"uri":"rtmp://myserver.com/live/myStream",
"localStreamName":"stream1",
"hasAudio": false,
"hasVideo": true
} |
Configuration
Codecs parameters definition
In the /usr/local/FlashphonerWebCallServer/conf directory you can find the SDP description file for the RTMP agent rtmp_agent.sdp :
| Code Block | |||
|---|---|---|---|
| |||
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 m=audio 0 RTP/AVP 103 96 97 98 99 100 102 108 104 a=rtpmap:108 mpeg4-generic/48000/12 a=rtpmap:96 mpeg4-generic/8000/12 a=rtpmap:97 mpeg4-generic/11025/12 a=rtpmap:98 mpeg4-generic/12000/12 a=rtpmap:99 mpeg4-generic/16000/12 a=rtpmap:100 mpeg4-generic/22050/12 a=rtpmap:104 mpeg4-generic/24000/12 a=rtpmap:102 mpeg4-generic/32000/12 a=rtpmap:103 mpeg4-generic/44100/12 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
| Code Block | ||||
|---|---|---|---|---|
| ||||
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:
| Code Block | ||
|---|---|---|
| ||
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
| Code Block | ||
|---|---|---|
| ||
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
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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:
| Code Block | ||
|---|---|---|
| ||
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
| Code Block | ||
|---|---|---|
| ||
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
| Code Block | ||
|---|---|---|
| ||
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
| Excerpt | |||||||||||||||||||||||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
1. A stream containing B-frames does not play or plays with artifacts (latencies, lags) Symptoms:
Solution
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:
Solution: use Fraunhofer AAC codec with the following parameter in flashphoner.properties file
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
This parameter also turns off AAC frames dropping. b) use Fraunhofer AAC codec
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. |
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