I was wondering if I can use an HTTP protocol to acquire an image stream from an RTSP camera? I am currently using VLC Media ActiveX Plugin to connect to and view the RTSP stream, but I would like to eliminate the ActiveX control and move to a more raw level of image acquisition. I recall seeing somewhere that it's possible to get these images using HTTP. I'd like to use Indy TIdHTTP component to connect to the camera and acquire the image. I'm also assuming this would need some sort of speed control, such as a delay in-between requests. However, it's also my understanding that these RTSP cameras have pre-defined frame rates, which using the standard RTSP protocol are supposed to follow.
many cameras will allow you to grab screenshots with a URL that might look like:
http://user:password#camera/snapshot.jpg
for a proper stream, you would need to use RTSP (there are Delphi RTSP clients), tunnelling over HTTP if your device supports the application/x-rtsp-tunnelled content type, or another stream your device supports.
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I'm using OpenCV to process a video stream that is provided via RTSP protocol. It's very easy to connect to the stream and read the frames. However, I'm having some timing issues and I want to make sure I'm doing it right.
The question is: how does VideoCapture::grab works in RTSP streams regarding frame buffering - does it retrieves the most recent frame provided by the stream or does it retrieves frames from a buffer? If so, can I control the buffer?
I am trying to stream RTP Packets (which is streaming an audio) from RTP URL e.g. rtp://#225.0.0.0
after so much research on the same i have somewhat streamed the URL in my device and playing it with https://github.com/maknapp/vlckitSwiftSample.
This is only playing the Streamed Data but does not have any function to store the data.
From research and other sources i dint find much content and simple information that should be helpful to stream the Packet over RTP and store it in iOS Device.
I have tried with following link.
https://github.com/kewlbear/FFmpeg-iOS-build-script
https://github.com/chrisballinger/FFmpeg-iOS
These two are not even compiling due to POD Issues other projects or guide just giving me reference on RTSP Stream instead of RTP Stream.
If anyone can give us a guidance or any idea that how we can implement such things then it will be appreciated.
First foremost, you need to understand how this works.
The sender i.e. the creator of RTP stream is probably doing the following:
Uses a source for the data: In case of audio, this could be the microphone or audio samples or a file
Encodes the audio using a audio codec such as AAC or Opus.
Uses RTP packetizer to create RTP packets from encoded audio frames
Uses a transport layer such as UDP to send these packets
Protocols such as RTSP provides the necessary signaling information to provide better stream information. Usually RTP itself isn't enough as things such as congestion control, feedback, dynamic bit rate are handled with the help of RTCP.
Anyway, in order to store the incoming stream, you need to do the following:
Use a RTP depacketizer to get the encoded audio frames out of it. You can write your own or use a third party implementation. In fact ffmpeg is a big framework which has all necessary code for most of the codecs and protocols. However for your case, find a simple RTP depacketizer. There could be headers corresponding to a particular codec to make sure you refer to a correct RFC.
Once you have access to encoded frames, you can write the same in a media container such as m4a or ogg depending upon the audio codec used in the stream.
In order to play the stream, you need to do the following:
Use a RTP depacketizer to get the encoded audio frames out of it. You can write your own or use a third party implementation. In fact ffmpeg is a big framework which has all necessary code for most of the codecs and protocols. However for your case, find a simple RTP depacketizer.
Once you have access to encoded frames, use a audio decoder (available as a library) to decode the frames or check if your platform supports that codec directly for playback
Once you have access to decoded frames, in iOS, you can use AVFoundation to play the same.
If you are looking at an easy way to do it, may be use a third party implementation such as http://audiokit.io/
From what have gathered so far, Apple provided tools to make Mac to act as HTTP Live Streaming server. But my goal is different. I want to make iDevices to be the HTTP Live Streaming server. (for local network only)
Can it be done at all?
Yes and no. Apple does not provide a way to stream encoded media data, so that part is 100% up to you. Also, Apple does not provide a way to access encoded frames directly (i.e. you can easily get an encoded file or the raw frames, but not easily get "encoded frames'). So you need to develop a way to get these encoded frames from the files for streaming, or encode the raw frames on the fly.
It may or may not fit your use case, but if you first right the streamer portion, you should be able to say small/short clips to disk, and stream them out as they are created with minimal overall latency.
i am streaming some FTA channels from
http://www.tbsdtv.com/products/tbs6985-dvb-s2-quad-tuner-pcie-card.html
using mediaportal
http://www.team-mediaportal.com/
and then i get rtsp url from mediaportal of channel i timeshift
and vlc i can send that stream to mediaserver FMS to get HLS, HDS, RTMP, RTSP
i have 3 servers running erlyvideo (flussonic)
so it take care of the delivery.
i want some alternate solution beside that
i have done some methods to work this our
including
VLC
IPTVL
Dvbdream
but the quality is better when i stream some thing as file, only FMLE works good with live stream, but for that we only can use directshow enabled devices like
http://www.viewcast.com/products/osprey-cards
i am doing it on windows.
if some one have any more methods or want to share his version please do so
There is a WWW page with Flash stream on it. I want to download and forward this stream to another streaming server, when possible - replace audio stream (e.g. translate), but without recompressing video stream. Usual way for this ATM is to capture and broadcast Flash player view from the web page, which is obviously suboptimal because video needs to be recompressed, making the quality notably worse and loading the cpu.
Has someone an idea how to do it? VLC seems to be able making relay, but it also seems not to support RTMP at all.
if you're ready to do this programmatically you can use crtmpserver (C++) or red5 (Java) with any RTMP client, otherwise this question doesn't belong to SO