rtp

iOS - 直播流程，视频推流，视频拉流，简介，SMTP、RTMP、HLS、 PLPlayerKit

由 隐身守侯 提交于 2020-01-19 01:54:02
收藏笔记 1 . 音视频处理的一般流程： 数据采集→数据编码→数据传输(流媒体服务器) →解码数据→播放显示 1、数据采集： 摄像机及拾音器收集视频及音频数据，此时得到的为原始数据 涉及技术或协议： 摄像机：CCD、CMOS 拾音器：声电转换装置（咪头）、音频放大电路 2、数据编码： 使用相关硬件或软件对音视频原始数据进行编码处理（数字化）及加工（如音视频混合、打包封装等），得到可用的音视频数据 涉及技术或协议： 编码方式：CBR、VBR 编码格式 视频：H.265、H.264、MPEG-4等，封装容器有TS、MKV、AVI、MP4等 音频：G.711μ、AAC、Opus等，封装有MP3、OGG、AAC等 3、数据传输： 将编码完成后的音视频数据进行传输，早期的音视频通过同轴电缆之类的线缆进行传输，IP网络发展后，使用IP网络优传输 涉及技术或协议： 传输协议：RTP与RTCP、RTSP、RTMP、HTTP、HLS（HTTP Live Streaming）等 控制信令：SIP和SDP、SNMP等 4、解码数据： 使用相关硬件或软件对接收到的编码后的音视频数据进行解码，得到可以直接显示的图像/声音 涉及技术或协议： 一般对应的编码器都会带有相应的解码器，也有一些第三方解码插件等 5、播放显示： 在显示器（电视、监视屏等）或扬声器（耳机、喇叭等）里，显示相应的图像画面或声音

Rtp stream synchronization

由 眉间皱痕 提交于 2020-01-16 09:05:14
问题 I am working on a project where in I need to sync rtp streams. Here, I am making use of RTCP SR to synchronize.. Eg: For first stream I receive 6 rtp packets then I receive RTCP SR packet, as I know the time RTCP SR packet arrival, I subtract the RTP timestamp of this RTCP SR packet with the first rtp packets of stream and get the difference. Then I divide this by the difference which is monotonic between two rtp packets, say 160 for AMR and 320 for AMR-WB. Now, after the division I multiply

Looping an MP4 video

由 丶灬走出姿态 提交于 2020-01-16 07:49:11
问题 I need to interface with a piece of hardware that is expecting an MPEG-4 RTP stream from a camera (actually multiple streams from multiple different cameras). What we'd like to do is supply that video from a set of small .mp4 files, looped endlessly. 1 What I'm trying right now is to use libVLC in server mode, with the "--loop" argument. The code for this looks like the following: libvlc_vlm_add_broadcast(vlc, "test", ("file:///" + video).c_str(), "#rtp{dst=localhost,port=1234,sdp=rtsp:/

Video streaming over RTP using gstreamer

由 牧云@^-^@ 提交于 2020-01-06 07:24:42
问题 I am trying to stream a video file using gstreamer from one device to another over RTP. At the sender side I am using the following command : gst-launch filesrc location=/home/kuber/Desktop/MELT.MPG ! mpegparse ! rtpsend ip=localhost But this gives the following error : no element "rtpsend" , I downloaded all the rtp tools and still the same error. Am I using rtpsend in some wrong way? Also can someone give me the command line code for streaming video file(locally stored in my laptop and not

Decoding unix time stamp from extended RTP packet header to calculate latency

由 你离开我真会死。 提交于 2020-01-05 08:07:16
问题 I am working on a project where I am trying to calculate the latency of the packets received between two android devices using RTP. So I went on to extend the RTP header with a unix time stamp in it's 12th-19th bytes. I've received the packets now and tried to extract the unix time from them. However, I am doing something wrong in the decoding process as you can see in the screenshot. On the left, is the time I decoded from the packet, and on the right is the time of arrival. Please ignore

Decoding unix time stamp from extended RTP packet header to calculate latency

由 回眸只為那壹抹淺笑 提交于 2020-01-05 08:07:05
问题 I am working on a project where I am trying to calculate the latency of the packets received between two android devices using RTP. So I went on to extend the RTP header with a unix time stamp in it's 12th-19th bytes. I've received the packets now and tried to extract the unix time from them. However, I am doing something wrong in the decoding process as you can see in the screenshot. On the left, is the time I decoded from the packet, and on the right is the time of arrival. Please ignore

how to modify the following command line?

由 落爺英雄遲暮 提交于 2020-01-05 04:40:36
问题 I have the following gstreamer Command-line : gst-launch alsasrc ! mulawenc ! rtppcmupay ! udpsink host= 127.0.0.1 port=5555 It records Mono Voice and i can hear it ,if i listen on 5555 port ( echo IP was used). But i need to transmit Stereo . I have also tried my Microphone for stereo-recording Capability using the following command: arecord -vv -fdat voiceFile.wav and it works. Does anyone know how to specify stereo in the gstreamer command? 回答1: The problem is that rtppcmupay does not

RTCP receiver report sending interval

由 ≯℡__Kan透↙ 提交于 2020-01-04 13:39:03
问题 What is the sending interval of RTCP Receiver Report? In RFC 3550 I was only able to find computation the RTCP Transmission Interval for the server. But as a client I have no idea about members and senders (or do I?). So I'm a little bit confuse how to calculate the interval or should I send RTCP RR in periods or should I only send RR packet when SR is received? 回答1: The RTP and RTCP protocols don't make a distinction between client and server. Both are members within the RTP session and both

RTCP receiver report sending interval

由 混江龙づ霸主 提交于 2020-01-04 13:38:58
问题 What is the sending interval of RTCP Receiver Report? In RFC 3550 I was only able to find computation the RTCP Transmission Interval for the server. But as a client I have no idea about members and senders (or do I?). So I'm a little bit confuse how to calculate the interval or should I send RTCP RR in periods or should I only send RR packet when SR is received? 回答1: The RTP and RTCP protocols don't make a distinction between client and server. Both are members within the RTP session and both

RTCP receiver report sending interval

由 蓝咒 提交于 2020-01-04 13:38:27
问题 What is the sending interval of RTCP Receiver Report? In RFC 3550 I was only able to find computation the RTCP Transmission Interval for the server. But as a client I have no idea about members and senders (or do I?). So I'm a little bit confuse how to calculate the interval or should I send RTCP RR in periods or should I only send RR packet when SR is received? 回答1: The RTP and RTCP protocols don't make a distinction between client and server. Both are members within the RTP session and both
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