rtp

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 newbie with gstreamer and I am trying to be used with it. My first target is to create a simple rtp stream of h264 video between two devices. I am using these two pipelines: Sender: gst-launch-1.0 -v filesrc location=c:\\tmp\\sample_h264.mov ! x264enc ! rtph264pay ! udpsink host=127.0.0.1 port=5000 Receiver: gst-launch-1.0 -v udpsrc port=5000 ! rtpmp2tdepay ! decodebin ! autovideosink But with the first one (the sender) I got the following error: Setting pipeline to PAUSED ... Pipeline is PE*R*O L(LgIsNtG- l.a.u.n h-1.0:5788): CRITICAL

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: When I create an RTP packet in scapy with the following code, I get an error: "TypeError: clone_with() got multiple values for keyword argument 'payload'" from scapy.all import IP, UDP, RTP, Ether from scapy.utils import PcapWriter pktdump = PcapWriter("banana.pcap", append=False, sync=True) rtp = { "sequence": 1, "timestamp": 1, "marker": 1, "payload": 17 } pkt = Ether()/IP()/UDP(sport=12345,dport=12346)/RTP(**rtp) pktdump.write(pkt) but removing payload works. rtp = { "sequence": 1, "timestamp": 1, "marker": 1 } pkt = Ether()/IP()/UDP

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'm having some trouble figuring out how to create a simple rtp stream with gstreamer and display it on vlc. I've installed GStreamer 0.10.30 and VLC 1.1.3. My only requirement is to use MPEG4 or H.264 codecs. Right now, I can stream the GStreamer videotestsrc through this simple pipeline: gst-launch videotestsrc ! ffenc_mpeg4 ! rtpmp4vpay ! udpsink host=127.0.0.1 port=5000 which outputs the "caps" needed by the client to receive the stream: /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)video

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 currently have an Android application that is capturing video from the camera and streaming it over RTP. I do not use RTSP. I have created a SDP file that describes the RTP feed. I can open the SDP file in Quicktime 7 and can see the stream, attempting to open the SDP file in VLC does not work. I get the following error: live555 demux error: no data received in 10s, aborting I am monitoring the RTP packets using Wireshark and can see the packets on the network (see example below): 281956 2545.589171000 10.0.1.25 10.0.1.150 UDP 1442 Source

Does the HTML5 video element support non-HTTP-based (HLS, SmoothStreaming, etc.) live-streaming protocols? Does it support RTP/RTSP streaming protocols? Does it support RT M P? Are there specific browsers that support or don't support it? Jaruba HTML5 tag has very limited support on video sources. The video sources supported are also limited to what browser your visitors use. Please see: http://www.w3schools.com/html/html5_video.asp for a table of supported formats depending on browser. To sum it up, HTML5 Video supports MP4 on all browsers and OGG, WEBM in FireFox, Opera and Chrome. With that

I've been working on some streaming software that takes live feeds from various kinds of cameras and streams over the network using H.264. To accomplish this, I'm using the x264 encoder directly (with the "zerolatency" preset) and feeding NALs as they are available to libavformat to pack into RTP (ultimately RTSP). Ideally, this application should be as real-time as possible. For the most part, this has been working well. Unfortunately, however, there is some sort of synchronization issue: any video playback on clients seems to show a few smooth frames, followed by a short pause, then more

实时传输协议 （ Real-time Transport Protocol 或简写 RTP ）是一个 网络传输协议 ，它是由IETF的多媒体传输工作小组1996年在RFC 1889中公布的。 国际电信联盟 ITU-T也发布了自己的RTP文档，作为H.225.0，但是后来当IETF发布了关于它的稳定的标准RFC后就被取消了。它作为 因特网 标准在RFC 3550（该文档的旧版本是RFC 1889）有详细说明。RFC 3551（STD 65，旧版本是RFC 1890）详细描述了使用最小控制的音频和视频会议。 RTP协议详细说明了在 互联网 上传递音频和视频的标准数据包格式。它一开始被设计为一个 多播 协议，但后来被用在很多 单播 应用中。RTP协议常用于 流媒体 系统（配合RTSP协议），视频会议和 一键通 （Push to Talk）系统（配合H.323或SIP），使它成为 IP电话 产业的技术基础。RTP协议和RTP控制协议 RTCP 一起使用，而且它是创建在 UDP协议 上的。 中文名 实时传输协议 外文名 Real-time Transport Protocol 简 称 RTP 类 型 概念 Ŀ¼ 1 特征 2 组成 3 使用 4 报文格式 5 RTCP概要 6 封包结构 特征 编辑 实时传输协议（RTP）为数据提供了具有实时特征的端对端传送服务，如在 组播 或 单播

由于工作中需要用到rtp协议，java暂时没有比较好的开发框架，参考了其他的一些博文，自己摸索着使用jrtplib库，在此记录一下。 关于如何编译jrtplib库，在我的上一篇有讲解 jrtplib库移植到android上 可根据自身需求，参考实现的功能。 实现的功能 一、接收端 接收RTP数据，携带数据并回调通知Java端； 接收RTCP、BYE等数据，携带数据发送端IP字符串并回调通知Java端； 实现网络摄像机RTSP协议对接； 实现接收RTP数据，并转发RTP数据。 二、发送端 1、实现H264编码数据分包发送； 2、发送RTP数据； 3、接收RTCP、BYE等数据，并携带数据发送端IP字符串回调至Java。 准备工作 Android.mk 文件编写 LOCAL_PATH := $ ( call my - dir ) include $ ( CLEAR_VARS ) LOCAL_MODULE := jthread LOCAL_SRC_FILES := libjthread . a include $ ( PREBUILT_STATIC_LIBRARY ) include $ ( CLEAR_VARS ) LOCAL_MODULE := jrtp LOCAL_SRC_FILES := libjrtplib . a LOCAL_STATIC_LIBRARIES :=

2019独角兽企业重金招聘Python工程师标准>>> 实时传输协议（RTP）是一个Internet协议标准，它描述了程序管理多媒体数据实时传输的方式。最初在Internet工程任务组（IETF）的请求注解（RFC）1869中对RTP协议进行了描述，RTP由IETF的音视频传输工作组设计，它支持多个地域上分布的参与者的视频会议。RTP普遍应用于Internet的电话应用中。RTP本身并不保证多媒体数据的实时传输（因为这取决于网络特性），但是，当数据尽最大努力到达后它将提供必要的方法来管理这些数据。 　　RTP与控制协议（RTCP）配合工作，RTCP使得大的组播网络能够监视数据传输。监视能使接收器侦测到任何的包丢失，还可以补偿任何的延迟抖动。两个协议都独立于下面的传输层和网络层协议。RTP头中的信息将告诉接收器如何重建数据，并描述了比特流失如何打包的。通常，RTP工作于用户数据报协议（UDP）之上，但它也能使用其他的传输协议。会话发起协议（SIP）和H.232都使用RTP。 转载于:https://my.oschina.net/dminter/blog/205037 文章来源: https://blog.csdn.net/weixin_33757911/article/details/91888851

I have been trying to figure out how to stream my desktop (over LAN) using VLC and to achieve the lowest latency possible (<100ms). The goal is to have another computer receive the stream and potentially play games while streaming (i.e playing game from PC1 on the PC beside the TV). What settings should I use? I have tried multiple approaches but have yet to succeed. EDIT: I am open to using something other than VLC as well. I have also tried the same with VLC and couldn't ever get latency bellow 3 seconds. FFmpeg did wonders and finally provided a latency bellow 1 second. mpeg2video and UPD