RTSP authentication problem?
Hi,
I am playing with an RTSP camera (Tapo C210) that is on the local network.
The path to the stream is with credential obfuscated as and .
I tested the endpoint with VLC and it works and I can see the stream so the credentials are good.
I wanted to obtain some information from the camera so I wanted to get information about the session as per the RTSP documentation:...
rtsp://myadmin:mypassword@192.168.1.10:554/stream1
rtsp://myadmin:mypassword@192.168.1.10:554/stream1
myadmin
myadmin
mypassword
mypassword
Extending the jellyfish video room demo with a queue
Hey I am curious what a scalable way would be to add a queue in front of jellyfish rooms. I have a setup very similar to the jellyroom demo.
My current attempt is adding additional state for the queue to the
RoomService module and somehow using the max_children option of the DynamicSupervisor to add to the queue but its getting super convoluted to manage the state
Would creating another GenServer like RoomQueue to manage queueing rooms be a good idea? Any ideas would be appreciated...Debugging bundlex/unifex errors
Hello, I've been tinkering with membrane cross-compiled to Nerves (rpi4).
I've had features (e.g. microphone input) working recently but it seems it has broken with recent upgrades.
Would anyone have any tips on debugging the
unifex_create/3 error below?...RTP to HLS Disconnect and Reconnect Audio Stream
Hello everyone,
I currently have a microphone input which is sending UDP to a server and into a RTP input and finally streaming over HLS. The entire flow is working fine, however, I am trying to handle a scenario where the microphone gets disconnected and when it reconnects and starts sending UDP packets over to the server, I start receiving:
...
[warning] <0.2197.0>/:rtp/{:stream_receive_bin, 1}/:packet_tracker Dropping packet 39181 with big sequence number difference (-13014)
[warning] <0.2197.0>/:rtp/{:stream_receive_bin, 1}/:packet_tracker Dropping packet 39181 with big sequence number difference (-13014)
Custom RTC Endpoint
Sorry, more random questions!
What's the best place to start when trying to get a custom endpoint to stream mpegts video in real-time to the browser?
I have a working pipeline for mpegts in a UDP stream rendering in the browser via HLS - basically UDP -> MpegTS -> Demuxer -> H264 Parser -> HttpAdaptiveStream. My application is real-time so the HLS lag is too much so WebRTC seems like the way to go....
HTTPAdaptiveStream issue with hls.js
I was wondering if anyone else has experieneced this issues with HLS?
I have a working pipeline generating HLS output which plays fine with Safari and it's native support for HLS.
However, when trying to get playback working in Chrome using hls.js it seems to stick at the buffering stage. It downloads the index.m3u8 file which looks like this:...
Spinning up a new GenServer for each room
I have been learning from the videoroom demo and I have a few questions.
```elixir
meeting.ex
@impl true...
RTP stream
I'm trying to get a simple RTP stream pipeline working but hit the following error:
```
16:22:57.782 [error] GenServer #PID<0.308.0> terminating
** (KeyError) key #Reference<0.1227858978.1981546498.158967> not found in: %{}
:erlang.map_get(#Reference<0.1227858978.1981546498.158967>, %{})...
React-Native connection?
I'm struggling to get a react-native client to connect to my membrane server. I'm just running locally right now. I start my membrane server with EXTERNAL_IP={my ip} mix phx.server. I'm using the
@jellyfish-dev/react-native-membrane-webrtc client in my react native code
Then I have the following connection code in my react-native view. I see the console log statements for init connect and attempting connect. But it never connects. I don't see a connection message in my phoenix server, nor the successful connection message.
I tried increasing the log verbosity, but didn't get anything out of the logs from react-native. Is there something obviously wrong with my connection string? Is it expecting something different for the server URL?...RTP demo with RawAudio
Hello friends, I'm trying to get microphone input (via
Membrane.PortAudio.Source) packaged into an RTP stream and sent to a server and can't quite seem to get it right.
Excerpt below based on the demo in membrane-demo/rtp but with microphone input substituted and newer syntax.
```...Unable to create new endpoints in Membrane RTC Engine 0.14.0
A change was made to the RTC engine, implementing a
to_type_string function for all existing endpoints. This function seems to be necessary for an endpoint to be added.
This has the side-effect of removing the ability to create new endpoints - only the predefined ones are allowed:
https://github.com/jellyfish-dev/membrane_rtc_engine/blame/master/lib/membrane_rtc_engine/endpoints/webrtc/media_event.ex#L366-L368
...How to pass some client side parameters to an RTMP pipeline
Hey team I have an RTMP pipeline (which I simplified for the purpose of the question):
```
def handle_init(ctx, socket: socket) do
Logger.info("Starting RTMP pipeline")...
How to generate SSL certs for membrane_rtc_engine dTLS?
Do you have any recommendations on how to obtain SSL certs for use in the handshake_opts of the WebRTC Endpoint?
Pipeline with RTMP source and AAC decoder
Hi everyone 🙂
(message can be redundant with the one I posted on slack, sorry about that)
I'm currently building a POC for live transcription using whisper model.
I checked Lawik example already which uses either the mic as or a file as a source to achieve it.
In my case I do have RTMP source with AAC codec for audio and flv format. ...
Intermittent Failures with RTMP Sink
We are running into some intermittent failures with the RTMP sink. What we notice is that sometimes a given pipeline will stream all the way through and sometimes the RTMP sink will raise when writing frames and crash the pipeline.
```
22:01:58.807 [error] <0.5365.0>/:rtmp_sink/ Error handling action {:split, {:handle_write, [[:video, %Membrane.Buffer{payload: <<0, 0, 139, 207, 65, 154, 128, 54, 188, 23, 73, 255, 152, 130, 98, 10, 43, 88, 28, 64, 176, 10, 39, 247, 233, 179, 54, 27, 17, 168, 97, 24, 82, 152, 175, 21, 138, 252, 216, 108, 205, 134, ...>>, pts: 27240000000, dts: 27240000000, metadata: %{h264: %{key_frame?: false}, mp4_payload: %{key_frame?: false}}}]]}} returned by callback Membrane.RTMP.Sink.handle_write_list
...
WebRTC to HLS, where does the pipeline happen?
Looking at the demo I can't see a "normal" pipeline. I want to process video frames in a pipeline. Do ML to them and stuff. I want the video from the browser. WebRTC should be a reasonable way to stream that. I want to apply a bunch of stuff between input and final HLS output.
Is there a straightforward way for that or is the engine mostly repackaging the video and audio streams?...
Membrane.Source example for RTC Engine
Hi! I'm trying to implement a solution in which I can stream audio from an API to the client via the Membrane RTC Engine.
I'm basing my Membrane.Source implementation on Membrane.Hackney.Source, but now I'm a bit stuck on getting the audio to send from that to the WebRTC Endpoint.
I've created an endpoint that fowards the output of my pad to Membrane.RTC.Engine.Endpoint.WebRTC.TrackSender, but I'm wondering:...
Problems specifying toilet capacity of Realtimer
I have a pipeline which I see fail intermittently on startup due to a toilet overflow of a realtimer element. The downstream element of the realtimer is a rtmp sink.
In my pipeline I have specified the toilet capacity with
via_in to the realtimer as shown in the docs:
```...Confusion on usage of MP4.Demuxer.ISOM
Hi, I'm trying to write a simple pipeline which take in an mp4 source and streams it out via the RTMP sink. I have some general confusion on how to properly wire up the MP4.Demuxer.ISOM.
1. How do I decode/parse the output pads stream format downstream from the source? I haven't seen any examples or demos of transforming the
MP4.Payload.{AAC.AVC1}
2. How to properly handle dynamically attaching the output pads for each track to downstream elements? If I want to handle the :new_track message to attach to some sink with pads in the :always availability mode (such as the RTMP sink) I can't attach that track to some grouping of elements which end at the sink temporarily. For example, if I get the :new_track notification for an AAC track I can't attach just the :audio pad of the RTMP sink, because when handling that callback there is no video pad to attach....PortAudio plugin on Apple Silicon Mac
Not sure what the issue stems from. Opening a microphone source gives me:
```
** (Membrane.CallbackError) Error returned from Membrane.PortAudio.Source.handle_prepared_to_playing:
:pa_open_stream...