Supporting interruptions in OpenAI Realtime demo
cancelled
):
```...membrane_rtc_engine/membrane_rtc_engine_ex_webrtc error
membrane_rtc_engine
with package membrane_rtc_engine_ex_webrtc
I'm seeing there's a dependency mismatch.
Ie; the example at the bottom of this page (https://github.com/fishjam-cloud/membrane_rtc_engine/tree/master?tab=readme-ov-file#repository-structure):
```...Demuxing Safari MP4
Membrane.MP4.Demuxer.ISOM
gives me an error:
```
Error parsing MP4
box: moof / traf / tfhd...SDL plugin fails to initialize
HTTP adaptive stream continuous segments
Sections of files
Pipeline Error: Pipeline Failed to Terminate within Timeout (5000ms)
burn in caption to mp4
stream RTMP to VLC Network stream
membrane_rtmp_plugin
to receive RTMP Stream as source (with help of Membrane.RTMPServer
and Membrane.RTMP.SourceBin
). All is fine, we migrated successfully to 0.26.0
version, which simplifies pipeline a lot. Also we did a POC of streaming RTMP to streaming service (Youtube) and everything is working as expected. I am curious, is there any way to stream RTMP to VLC Player (probably it is called pull approach)? I mean File -> Open Network -> Specify URL (eg. r...Fly.io + UDP
url
in that message specifies UDP as the transport. I deployed to fly.io, and explicitly did not open up the UDP ports in my fly.toml, so I'm wondering why the app is failing with a UDP timeout. Am I incorrect in assuming that I can force all traffic over TCP by just not opening it up? Shoudl I also figure out UDP?
On UDP, I read through this: https://github.com/fishjam-dev/fishjam-docs/blob/main/docs/deploying/fly_io.md
But it's Fishjam specific, and it doesn't line up neatly with my app which is based on the old membrane video room repo (https://github.com/membraneframework-labs/membrane_videoroom). Where does fly-global-services get specified in that case? I'm not explicitly setting a TURN_LISTEN_IP. I traced through things I think it could be here in turn_ip
(and then the turn_mock_ip
is my external IPv4 address)...web rtc engine and erlang clustering / load balancing
WebRTC TURN TCP/TLS configuration issue
Syncing two streams from HLS source
demuxer2
is the audio stream which seems to produce buffers at a much lower rate. ...Lowest latency h264 UDP video stream possible.
Using Google meet as a source
Background loop with sound effect playback on event
Retransmit received RTP packets in secure way
membrane_webrtc_plugin: %Membrane.Buffer with pts: nil, dts: nil received from audio track.
membrane_webrtc_plugin
to connect the streamer and the viewers.
Everything is working pretty fine with video track. However, when I'm adding the audio one I'm starting to receive a bunch of errors, namely:
1) ArgumentError from membrane_realtimer_plugin, handle_buffer/4 function where it essentially tries to do subtraction from nil:...Pipeline stuck at MP4 Demuxer
how to create mp4 file chunks with File.Sink.Multi and ISOM