It is possible to decrypt an encrypted video in AES/CTR and and watching as it was being decrypted?
I want to decrypt Video hosted on MEGA.co.nz and go watching as that goes down without waiting for the download to complete.
I already know how to decrypt, just need to know how to do this gradually in a video
Yes, it's possible. Most video formats and codecs can be decoded in a streaming fashion. For example: MPEG4. AES in CTR mode can be decrypted as a stream.
Related
I'm trying to decrypt UDP packets for a multiplayer video game. When loading into a game session, a DTLS handshake occurs where, in Wireshark, I usually see the Client and Server agree on ChaCha20 Poly1305 encryption. The game actually live logs a "key" in a log file, which is 32 bytes long hex-coded, along with an HMAC and IV. At this point I'm not sure what to do. I tried decrypting individual messages in Python with some cryptography libraries but I realized that might be silly upon learning DTLS, or at least TLS packets, cannot be decrypted independently. I know I can possibly have Wireshark point to a file or add a key to live decrypt something, but have not had luck doing so. I started this process from basically no knowledge on internet security protocols or cryptography and have learned a lot but am at a standstill, and just want to make sure I'm not far off-base here.
Wireshark screenshot of handshake
It depends on what the game is actually writing to the file. Wireshark has support for decrypting TLS/DTLS using the RSA private key, the premaster secret or master secret. If the log file contains the premaster or master secret, then you should be able to shoe-horn it into wireshark, and decrypt the stream from there.
If it isn't, then you'll need to work out what it actually is first, and then it's a bit more of a manual job to get at the data.
As i starting to work with video streaming, i've got a question:
Video streaming is the process of breaking video file into small data packages that are sent over network. But where do they stored and what happen with it after streaming was finished? I am asking because unlike from download, streaming does not keep the file locally, that's how it described in internet. What is the process of handling stream buffers under the hood. Can someone point me into right direction?
Any help appreciated
Thanks
Most video streams are actually HTTP request and response based - i.e. he client (player) request the video chunk by chunk and then plays it as it receives each chunk.
To answer your question what happens to the chunks when they are downloaded, this will depend on the player and the device. In general the chunks will be rebuilt into the particular video container that is being used, e.g. mp4, and then played.
How long they are stored will depend on the device and the players caching rules and capacity.
I was hoping to build an application that streams audio (mp3, ogg, etc.) from my microphone to a web browser.
I think I can use the html5 audio tag to read/play the stream from my server.
The area I'm really stuck on is how to setup the streaming http endpoint. What technologies will I need, and how should my server be structured to get the live audio from my mic and accessible from my server?
For example, for streaming mp3, do I constantly respond with mp3 frames as they are recorded?
Thanks for any help!
First off, let's split this problem up into a few parts. You have the audio capture (recording), the encoding/codec, the server, and the receiving clients.
Capture -> Codec -> Server -> Several Clients
For audio capture, you will need to use the Web Audio API along with getUserMedia. This will allow you to get 32-bit floating point PCM samples from the recording device. This data stream takes up a ton of bandwidth... a few megabit for a stereo stream. This stream is not directly playable in an HTML5 audio tag, and while you could play it on the receiving end with the Web Audio API, it takes up too much bandwidth to be useful. You need to use a codec to get the bandwidth usage down.
The codecs you want to look at include MP3, AAC (and its variants such as HE-AAC), and Opus. Not all browsers support all codecs. MP3 is the most widely compatible but AAC provides better quality for a given bitrate. Opus is a free and open codec but still doesn't have the greatest client adoption. In any case, there isn't yet a codec that you can run in-browser with any real stability. (Although it's being worked on! There are a lot of test projects made with Emscripten.) I solved this problem by reducing the bit depth of my samples to 16-bit signed integers and sending this PCM stream to a server to do the codec, over a binary websocket.
This encoding server took the PCM stream and ran it through a codec server-side. Here you can use whatever you'd like, such as a licensed codec binary or a tool like FFmpeg which encapsulates multiple codecs.
Next, this server streamed the data to a real streaming media server like Icecast. SHOUTcast and Icecast servers take the encoded stream and relay it to many clients over an HTTP-like connection. (Icecast is HTTP compliant whereas SHOUTcast is close but not quite there which can cause compatibility issues.)
Once you have your streaming server set up, it's as simple as referencing the stream URL in your <audio> tag.
Hopefully that gets you started. Depending on your needs, you might also look into WebRTC which does all of this for you but doesn't give you options for quality and also doesn't scale beyond a few users.
After reasearching for a few days, i m still lost with this issue:
I have a webcam connected over WiFi to my Android device.
I wrote an Android app to connect to a specified Socket of the webcam (IP and port). From this Socket i get an InputStream which is already encoded in H.264. Then i redirect this InputStream from the android device to my server, where i managed to decode it to images/frame by using Xuggler.
I would like to stream my webcam live to the internet to a flash player or something.
I know i have to use Wowza, FMS or RED5 for this.
My problem is, that i dont understand how to proceed with the InputStream i have. All examples i ve read need a mp4/flv or other container file to stream from... but i have a continuous live InputStream.
Some other examples consider using Flash Encoder. But my InputStream is already encoded in H.264.
This is a general understanding question. Please advise me on how to solve this.
Thank you
you have following options -
Encode in flv container. Yes you can transmit live stream using using flv container. You can set the 'duration' field in the header to be arbitrary long. e.g youtube use this trick for live streaming.
you can encode the stream into RTMP. ffmpeg has code for rtmp code which can be used for understand, or i believe there are other opensource rtmp muxers available.
convert the stream into HLS, there are flash based HLS player available.
why flash if I may ask, hope you know that HTML5 video tag now directly accepts h264 encoded videos.
I have a video that needs to be delivered through streaming, but all viewers need to be synchronized at the same time regardless of when they started the video. If the video starts streaming at 7:00 and someone visits the page at 7:05, they should see the footage at 7:05 and onwards.
Does Red5 or Flash Media Server or any other streaming server have a feature to handle this? or is this something that needs to be handled by the player?
regardless of how you load an active stream in Flash, it will start at the beginning of the file stream. For real-time streams that is the moment the user joins the stream since the file stream starts at that moment.