Search results

We run Flashphoner with CDN 2.0. with 1 Origin, and 1-5 edges. Edges servers are regularly added and removed. We have noticed that even once an edge server has been removed, the origin will continue to try to connect to it, resulting in wasted resources. Is there a way to set a timeout on edge...

Appears to be working now. Thanks.

We will test and let you know.

Tested on Flashphoner 3496 with the following config: record_rotation=120 webrtc_cc2_twcc=false Recording still appears to be broken. We ran a stream for 112 minutes, and recorded 2007 files (new file every ~3.3 seconds).

We will test and let you know.

Just sent logs in.

Tested and confirmed this is fixed. Thank you.

Tested on 3482 and it is still broken. We setup 2 brand new servers to test with. The only config option that we used was: record_rotation=120 Besides that, all other config files are using default settings. There was 1 stream broadcast to each server, and there were no subscribers. We broadcast...

We will start up a new test on the latest Flashphoner build and let you know what we observe.

When recording with Flashphoner and using file splitting, the server sometimes will start spitting out unusable files every couple seconds. We have observed this issue since ~5.0.31xx, up until the newest version of 5.1.3415. Sometimes it takes a couple minutes for this issue to happen...

This still appears to be broken on the latest version. I just ran a test on 5.1.3415-563991fd257ef8a79277bc5bb903766d987d2ef2 The following config options were set in flashphoner.properties stream_record_policy=template stream_record_policy_template={streamName}${mediaSessionId}...

Thank you. I will test it and see if that fixes it. Is there any way to patch it on the older version 5.0.2971?

When broadcasting WebRTC audio only stream, and using the flashphoner.properties.record_rotation=600, the recording stops after 5 minutes and does not start a new file. This happens to us on the latest Flashphoner 5.1.3376-5180923ad12b54518a270710810d6c886d27f067, as well as on versions as old...

When starting up the broadcast, the quality is often very pixelated and bandwidth usage is low. Over the span of about 25 seconds, the stream settles and finally begins broadcasting at a constant, high-quality level. * This is with Chrome on Windows. We've looked into Chrome-related options and...

When will an updated AMI be available for the AWS Marketplace? We would like to launch C5 or M5 instances, but the existing AMI is not compatible due to ENA being disabled: FlashphonerWebCallServer-5.0.2106-x86_64-319e70db-b450-4c78-b2d5-cccbf7c4efa0-ami-1bd70e0d.4 - ami-1ddb030b...

If we use the recording option, it creates a series of recording files based on the start time of the first recording. When the recording rotation (5 minutes) occurs, it uses the start time for the recording session and not the time that file begins recording. Is there a recording filename param...

I have an audio/video WebRTC stream that I recorded yesterday. The actual broadcast/recording was 3 hours 57 min 31 sec. When I load up the recording for playback, it shows a length of 16 hours 34 min 37 sec. In addition, the video stops about 35 minutes before the end of the file (Audio is...

We have been using audio-only RTC streaming for a while, but a recent release breaks this functionality. We can reproduce this on your official demo server: https://wcs5-eu.flashphoner.com/client2/examples/demo/streaming/media_devices_manager/media_device_manager.html Google Chrome: Uncheck...

From previous threads, it does not appear to be an option to broadcast with FMLE VP6 video and have that transcoded to a format that is supported with WebRTC playback. Can you confirm this is still the case? Regarding audio, FMLE on Windows (without the additional AAC encoder) only supports MP3...

I removed the wcs db files before running the wcs updater. That did not work I also configured the server with mysql. I've dropped the wcs schema and recreated it. That also did not work