使用jrtplibRTP传输H264视频文件 CSDN博客
Realtime Transport Protocol Wikipedia
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Why WebRTC chose RTP max packet size to 1200 bytes
This is an arbitrarily selected value to avoid packet fragmentation There is no any exact science behind this as you can be never sure on the actual limits however 1200 byte is a safe value for all kind of networks on the public internet including something like a double VPN connection over PPPoE and for RTP there is no much reason to choose a bigger value
上一篇我们介绍了RTP协议的一些基本知识下面我们介绍如何使用jrtplib这个库传输H264编码cpp view plaincopy JRTP传输 好了下面是我写的使用JRTP进行发送H264数据包的例子具体解释可以看注释发送端也可以接收接收端发送过来的RTCP数据包define MAXRTPPKTLEN
Is there a way to adjust the distance rtp takes me and how to turn back on because i think the limit is at like 1000 blocks and i want it to be way further cos i just keep rtping into places iv been i sometimes rtp 10 blocks away Max rtp distance Drtpmaxdistance20000 Increase this to what you want
SDP中H264 RTP Payload各项参数含义 CSDN博客
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The Catfather RTP Confusion Need Clarification
The Realtime Transport Protocol RTP is a network protocol for delivering audio and video over IP networksRTP is used in communication and entertainment systems that involve streaming media such as telephony video teleconference applications including WebRTC television services and webbased pushtotalk features RTP typically runs over User Datagram Protocol UDP
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丢失了部分rtp协议数据包造成的影响是可能是马赛克 卡顿或者跳帧max delay reached need to consume packet 表示这是 RTSP协议处理器报告的信息后面的十六进制数是处理器实例的内存地址max delay reached 表示达到了最大允许的延迟时间
ffmpeg拉取rtsp网络视频流报错解析max delay CSDN博客
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RTP和RTCP RTP全名是Realtime Transport Protocol实时传输协议它是IETF提出的一个标准对应的RFC文档为RFC3550RFC1889为其过期版本RFC3550不仅定义了RTP而且定义了配套的相关协议RTCPRealtime Transport Control Protocol即实时传输控制协议RTP被定义
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