abouut rest api rtmp pull

[Max]

Good day.
You did not provide us with your settings file flashphoner.properties.
For faster resolution of the issue, please provide us SSH access to you server. Send authorization data to your server to email support@flashphoner.com.

 

[gurmeet]

# Config flashphoner.properties
# To get more settings:
# ssh -p 2001 admin@localhost
# default password: admin
# show node-settings
# show node-settings | grep port

#server ip
ip =109.237.25.18
ip_local =109.237.25.18

#webrtc ports range
media_port_from =31001
media_port_to =32000

#codecs
codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp =opus,g729,g722,mpeg4-generic,vp8,mpv

#websocket ports
ws.port =8080
wss.port =8443
transcoder_agent_activity_timer_timeout=600000
rtmp_transponder_full_url=true

 

[Max]

Good day.
We've checked your server, but there's no any stream on it. So, you should do the following:
1. Reproduce your problem
2. Collect report as described here and send us to support@flashphoner.com

 

[gurmeet]

thats y i asking that it is pull stream automatically close in one minute or about.
that is the problem. u can get live rtmp demo url from internet and test.
u understand what i mean to said

and also
setting this properties client_log_level=DEBUG
server not working properly

 

[gurmeet]

1. Post Url : https://tv1.crakex.in:8888/rest-api/push/startup

2. Body Section :

{
"streamName": "rtmp://13.233.141.185:5119/live/ub2020",
"rtmpUrl": "rtmp://localhost:1935/live/ub2020"
}

3. JSON Response :

{
"exception": "com.flashphoner.rest.server.exception.NotFoundException",
"path": "/rest-api/push/startup",
"error": "Not Found",
"message": "Can not start transponder",
"timestamp": 1565693766219,
"status": 404
}

Thank you.

 

[Max]

{
"streamName": "rtmp://13.233.141.185:5119/live/ub2020",
"rtmpUrl": "rtmp://localhost:1935/live/ub2020"
}

You cannot pull stream and push stream in the same REST query. You should do the following:
1) Pull stream

/rest-api/pull/rtmp/pull
{
"uri": "rtmp://13.233.141.185:5119/live/ub2020"
}

2) Push stream

/rest-api/push/startup
{
"streamName": "rtmp://13.233.141.185:5119/live/ub2020",
"rtmpUrl": "rtmp://localhost:1935/live/ub2020"
}

3) Then you can play stream pushed using WebRTC for example

Also, you shoud set the following parameters in flashphoner.properties file

rtmp_transponder_full_url=true
rtp_activity_detecting=false,60

for this to work.

 

[gurmeet]

plz check this link
http://205.147.102.6/g/sites/test/record.mkv

 

[Max]

The RTMP stream in the recording has no audio, therefore it fails on RTP activity after a minute.
Add to flashphoner.properties

rtp_activity_detecting=false,60

and restart the WCS to apply the setting.
Then RTP activity detection will be disabled, and stream will not fail.

 

[gurmeet]

its taking too much time to load the page
browser (panel) responding very slow
Thread(s) per core: 1
Core(s) per socket: 1
Socket(s): 6
NUMA node(s): 1
6 cores cpu
32 gb ram
wcs responding very slow
video in ios safari not working
ios chrome also not working

 

[Max]

In case there is a performance question, please create a new forum thread. This thread is about pulling RTMP.
The performance issue has not been reproduced with the server, which credentials you sent previously. Please clarify – in a new thread – if it occurs with a different server, and if that is the case send that server credentials.

 

[gurmeet]

i allready sended to support mail
video in ios safari not working
ios chrome also not working

 

[Max]

"OutOfMemoryError: Java heap space" in server log - increase maximum and initial heap values in WCS_HOME/conf/wcs-core.properties.
Replace

-Xmx1024M

with, e.g.

-Xms3g
-Xmx3g

and restart the server.