rtp transmission

Video courses and related documents code address: https://github.com/EasyDarwin/Course#course-3RTP ProtocolThe real-time Transport protocol RTP (real-time Transport Protocol) is a network transport protocol that was published by the IETF Multimedia Transmission Working Group in RFC 1889 in 1996 and later updated in RFC3550.ITU-T also released its own RTP document

RTP (Real Time Transport Protocol) RTP is a transmission protocol for multimedia data streams over the Internet.Rfc1889Release. RTP is defined to work during one-to-one or one-to-many transmission. It aims to provide time information and implement stream synchronization. Ty

coding standards jointly developed by a Joint Video group (JVT) consisting of the ITU-T video coding Expert Group (VCEG) and the ISO/IEC dynamic image Expert Group (mPEG, its biggest advantage is its high data compression ratio. h. 264 of the compression ratio is more than 2 times of the MPEG-2, is the MPEG-4 of 1.5 ~ 2 times. At the same time, the video encoding layer (VCL) and network extraction layer (NAL) are used.The hierarchical design is very suitable for real-time

Http://blog.sina.com.cn/s/blog_488365030100ccp8.html 1Real-time audio and video transmission 1.1JrtplibLibrary Introduction The system uses the open-source database jrtplib to develop the RTP transmission module. Jrtplib is a fully open-source RTP Library developed by EDM (expertise centre for digital media) of Hasselt

Http://www.chinaitlab.com/cisco/RIP/832426.html1. IntroductionAt present, the realization of real-time voice, video communication and application in IP network has become a mainstream technology and development direction of network application, this paper introduces the standard protocol RTP for real-time voice and video data transmission in IP protocol family (real-time Transport Protocol) and RTCP (

Real-time transmission protocol RTP and RTCP Author: Streaming Media Chinese network released streaming media Chinese network provided 3:46:00 RTP (Real-timetransportprotocol) is a transmission protocol f

I was recently depressed by the load type and timestamp of RTP. After debugging for nearly a week, I finally solved the problem. Let's look back, I found that the main reason is that I did not really understand the meaning of the load type and timestamp in the RTP protocol. Although RTP transmission is supported by Jrt

Implementation principle: RTP protocol-based video transmission system: Principle Related articles: "1" RTP protocol analysis Introduction to "2" jrtplib "3" QT call Jrtplib for unicast, multicast, and broadcast "4" RTP Payload (payload) type, RTP Payload type Production of

1. Applicable H.264 Video Transmission Mechanism RTP is discussed earlier. Protocol and the basic stream structure of H.264, how can we use RTP to transmit H.264 videos? An effective method is to strip each NALU from the H.264 video, add the corresponding RTP Header before each nalu, and then send packets conta

Real-time transmission protocol (RTP) provides end-to-end transmission services with Real-Time Characteristics for data, such as interactive video audio or analog data under multicast or Unicast Network Services. Applications usually run RTP on UDP to use its multi-node and verification service. Both protocols provide

RTP/RTCP/RTSP/SIP/SDP relationship1. RTPReal-time Transport Protocol is a Transport layer protocol for multimedia traffic on the Internet. The RTP protocol details the standard packet format for transmitting audio and video over the Internet. RTP protocols are commonly used in streaming media systems (with the RTCP protocol), video conferencing and a Push-to-talk

1. RTP speex Header The RTP Header is defined in [rfc3550. This document defines the usage of fields in the RTP Header. Payload type (PT): the charge type number in this format. Marker (m) bit: this bit is used to mark the beginning of a silent sound. Place it on the first package of the audio data. Speex supports sound detection and does not generate frame da

Real-time audio and video domain UDP is the king In the Internet, audio and video real-time interaction using the Transport Layer Scheme has TCP (such as: RTMP) and UDP (such as: RTP) two kinds. The TCP protocol can provide a relatively reliable guarantee for data transmission between two endpoints, which is achieved through a handshake mechanism. When the data is passed to the receiver, the receiver checks

streaming media is not: for RTSP, and HTTP as normal communication, the key is for RTP packets, the server is in the NAT inside the player, even if RTSP know the NAT inside the player host address and NAT address, RTP packets to reach N At, the above 2 situation, because the mapping relationship has not been established before, this packet will be discarded by NAT. Some streaming media server will use HTTP

real-time audio and video domain UDP is the kingly The Transport layer scheme for real-time audio and video interaction on the Internet has two types: TCP (e.g. RTMP) and UDP (e.g. RTP). The TCP protocol provides a relatively reliable guarantee for data transmission between two endpoints, which is achieved through a handshake mechanism. When the data is passed to the receiver, the receiver checks the correc

();Recvoffset = 0; // After uploading, initialize// Printf ("Debug... 3/N ");}Return flage;} Summary: RTP/RTCP data transmission process: Server: Send the fixed-length data to the client in batches and send it to the client in the form of packet. That is to say, a packet needs to be sent several times by packet. After the packet is sent successfully, the next packet always calls the function: Session. sen

The Wgscd-picked Rtp/rtcp (real-time transport protocol/real-time Transport Control Protocol) is based on UDP-derived protocols and adds control over real-time transmission. Commonly used for online transmission of real-time video data, such as remote video surveillance, video-on-demand. There is a book called "Multimedia Network

UDP-based RTP transmission in the complex public network environment, especially 3G, 4G, WiFi network faced with packet loss, disorderly sequence, repetition, jitter and other issues, seriously affect the real-time audio and video interactive effect, even if it is a RTP packet lost, if the receiver does not do processing, will also lead to the appearance of video

IX h264 RTP transmission details (1) In the previous chapters, the introduction to the server has a very important issue that has not been carefully explored: how to open a file and obtain its SDP information. Let's start from here. When the rtspserver receives a DESCRIBE request to a media set, it finds the corresponding servermediasession and calls servermediasession: generatesdpdescription (). In genera

When using RTP to transmit H264 data, when the length of the Nalu is too long to subcontract, here is an example, if you want to know more detailed protocol description can refer to the end of the connection. Within live555, receive a piece of data at the beginning of each package as follows 7c Bayi E1 427c 1 D 8f7c 1 A7 C87c 1 2d。。。7c 1 6b FB7c 1 2b7c 3b live555 the above data processing, the resulting data is returned to the user as follows E1, 7f,