Try this config. From our testing, we found the IP resolving never worked, and wasn't even needed.
Edge
cdn_enabled=true
cdn_ip=0.0.0.0
cdn_point_of_entry=origin-server.foobar.com
cdn_role=edge
Origin
cdn_enabled=true
cdn_ip=0.0.0.0
cdn_role=origin
We ran tests on WCS 5.2.56
Config:
/usr/local/FlashphonerWebCallServer/conf/flashphoner.properties
stream_record_policy=template
stream_record_policy_template={startTime}_{endTime}_{streamName}_{mediaSessionId}_{audioCodec}_{videoCodec}
record_rotation=300
The files should be splitting...
We saw errors with the modified time when the stream had a poor connection and was stopped by the server. The actual file could contain ~10 seconds of media, but the last modified would be ~25 seconds in.
Could that part start / end time be added to the filename template so that we can avoid...
I tested on WCS 5.2.36, the startTime is just the stream start time, not the file start time.
In previous builds we have tested using the modified at time, but it appears WCS sometimes touches the file after the stream has finished.
We are using
record_rotation=300
For each of the files that are created, we need to know the timestamp that that recorded part was started at, and the timestamp it ended it.
For example:
Config:
record_rotation=300
stream_record_policy=template...
Is there an estimated completion time for this?
We've attempted to use the file modified at time, but sometimes it seems Flashphoner touches the file after it is finished, so we can't get an accurate timestamp.
Yesterday (3/6/19) we experienced streams randomly going black & no audio. On the V1 server, when we tried to restart the broadcast we got a 'stream already published' error. On the V2 server, we were able to restart the broadcast.
Not all the streams on the server went black. We weren't able...
What is the easiest way to collect a list of all stream parts? We'd like to include the start and end timestamps for the recorded portion. We've tried to generate a start/creation timestamp using the file system, but it is unreliable. We have tried to use the modified time from the file system...
When broadcasting audio / video on a low bandwidth connection, is there any way to prioritize the audio stream over the video stream? So in the case that the bandwidth drops, the video with drop packets, but the audio will not?
We are using a multi-origin setup for broadcasting. Is there any easy way to take a screenshot of a stream, without having to lookup which origin server the stream is being broadcast from?
We had this stun/buffer issue again today. Running WCS.version 5.1.3592-562aac10c6e8144f491da9a9393ff5b0f9be532c
This was in our origin/edge setup. Restarting the origin server resolved the issue, but we were unable to start new broadcasts until a restart.
FP: v. 0.5.28.2747 - 5.1.3592-562aac10c6e8144f491da9a9393ff5b0f9be532c
Chrome: Version 70.0.3538.67 (Official Build) (64-bit)
When specifying the audio bitrate, the following error is thrown:
This only occurs in the latest version of Chrome.
A temporary fix is to not specify the audio...
Recently we ran into an issue with our CDN 2.0 setup.
Sometimes when starting the stream, the server will begin dumping:
02:59:59,881 ERROR StunDatagramSocket - Stun receiver udp/3xxxx Incoming buffer exhausted!
After this incoming streams will not broadcast.
We are doing further...
Is there any documentation available about the capacity of Flashphoner? We are broadcasting the following stream:
Resolution: 1280x720
Framerate: 15fps
Bitrate: 1000kbps
Recording: true
In our testing, we are looking at approximately 150 streams / server before we begin to have stream...