rtp

I'm trying to encode and stream using ffmpeg (libavcodec/libavformat - MSVC x64 with Zeranoe builds) Here is my code, largely adapted from the encoding example, error handling removed #include "stdafx.h" extern "C" { #include <libavformat/avformat.h> #include <libavcodec/avcodec.h> #include <libavutil/opt.h> #include <libavutil/channel_layout.h> #include <libavutil/common.h> #include <libavutil/imgutils.h> #include <libavutil/mathematics.h> #include <libavutil/samplefmt.h> } #pragma comment(lib, "avformat.lib") #pragma comment(lib, "avutil.lib") #pragma comment(lib, "avcodec.lib") int main() {

I have a read/seek input stream of a video file (.avi mpeg4\xVid\ect..) in C# and I would like to stream it to a video player with jump to moment X feature enabled. How can I implement this? I heard that RTP might be a good protocol. What I'm really looking for is a library in C# that will help me out. Thanks in advance. Yes, Streamcoders solution is very good if you have some €1890. But if you do not want to pay money you should look at some free libraries or write your own from scratch. Take a look at these articles with C# sources: RTP VOIP Library Project How to use the managed RTP classes

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I am trying to capture the stream of two IP cameras directly connected to a mini PCIe dual gigabit expansion card in a nVidia Jetson TK1. I achieved to capture the stream of both cameras using gstreamer with the next command: gst-launch-0.10 rtspsrc location=rtsp://admin:123456@192.168.0.123:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink rtspsrc location=rtsp://admin:123456@192.168.2.254:554/mpeg4cif latency=0 ! decodebin ! ffmpegcolorspace ! autovideosink It displays one window per camera, but gives this output just

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I want to broadcast a video file via FFmpeg, but I get this error: Only one stream supported in the RTP muxer I get that error when I write this: ffmpeg.exe -i SomeVideo.mp4 -f rtp rtp://127.0.0.1:11111 I don't know what's wrong. 回答1: Your ffmpeg command creates two streams, one for video, one for audio. Do this instead: ffmpeg -re -i SomeVideo.mp4 -vcodec copy -an -f rtp rtp://127.0.0.1:11111 -vn -acodec copy -f rtp rtp://127.0.0.1:11112 Port 11111 then has video without audio ( -an ). Port 11112 then has audio without video ( -vn ). Read

I've managed to set up a SIP call using the JAIN-SIP API for Java. Now I would like to stream some video once a dialog has been established. I've read that this is possible with SDP and RTP, and I've found multiple examples on how to define a SDP/RTP body in a SIP packet. But once you have negotiated capability etc. on nodes, how do you actually start the RTP stream? Do you start an RTP streaming server outside or inside your Java application? If so, how? What is the link? In what I'm able to find online, nodes just "start exchanging RTP packets". Thank you. You need an RTP stack. As you are

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I need to detect MPEG4 I-Frame in RTP packet. I know how to remove RTP header and get the MPEG4 frame in it, but I cant figure out how to identify the I-Frame. Does it have a specific signature/header? 回答1: Ok so I figured it out for h264 stream. How to detect I-Frame: remove RTP header check the value of the first byte in h264 payload if the value is 124 (0x7C) it is an I-Frame I cant figure it out for the MPEG4-ES stream... any suggestions? EDIT: H264 IDR This works for my h264 stream ( fmtp:96 packetization-mode=1; profile-level-id=420029

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I am using kurento's master git to make a WebRTC to RTP bridge. MediaPipeline pipeline = kurento.createMediaPipeline(); WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build(); HttpGetEndpoint httpEndpoint=new HttpGetEndpoint.Builder(pipeline).build(); org.kurento.client.Fraction fr= new org.kurento.client.Fraction(1, 30); VideoCaps vc= new VideoCaps(VideoCodec.H264,fr); httpEndpoint.setVideoFormat(vc); AudioCaps ac= new AudioCaps(AudioCodec.PCMU, 65536); httpEndpoint.setAudioFormat(ac); webRtcEndpoint.connect

I'm having some trouble streaming H.264 video over RTSP. The goal is to live-stream a camera image to an RTSP client (ideally a browser plugin in the end). This has been working pretty well so far, except for one problem: the video will lag on startup, stutter every few seconds, and has a ~4-second delay. This is bad. Our setup is to encode with x264 (w/ zerolatency & ultrafast) and packed into RTSP/RTP with libavformat from ffmpeg 0.6.5. For testing, I'm receiving the stream with a GStreamer pipeline with gst-launch when connecting to an RTSP server. However , I've been able to reproduce the

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 由 翻译 强力驱动 问题: I get RTP stream from WebRTC server (I used mediasoup ) using node.js and I get the decrypted RTP packets raw data from the stream. I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other media servers. I guess that the best way would be to create SDP file that describes both the audio and video streams and send the packets therough new sockets. The ffmpeg command is: ffmpeg - loglevel debug - protocol_whitelist file , crypto , udp , rtp - re - vcodec libvpx - acodec opus -