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 the transport layer protocol function. However, RTP
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 the correctness of the data. The sender can
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
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 containing the
Streaming media refers to the continuous time-based media transmitted using stream technology in the network. It features that the entire file does not need to be downloaded before playback, but is played by downloading and playing, it is the technical basis for video conferences, IP phones, and other applications. RTP is a standard protocol and Key Technology for Real-Time Streaming Media transmission. This article describes how to program Real-Time
H.264 video RTP load format
1. Network abstraction layer unit type (NALU)
The NALU header consists of one byte. Its syntax is as follows:
+ --------------- +| 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 |+-+| F | NRI | type |+ --------------- +
F: 1 bit.Forbidden_zero_bit. The H.264 specification specifies that this digit must be 0.
NRI: 2 bits.Nal_ref_idc. 00 ~ 11. It seems to indicate the importance of this NALU. For example, the NALU decoder of 00 can discard it
In network transmission, let's cut down the RTP protocol. Let's take a brief look at some definitions and concepts of this Protocol. 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 R
Network Abstraction Layer Unit type (NALU):The Nalu header consists of a byte with the following syntax:+---------------+|0|1|2|3|4|5|6|7|+-+-+-+-+-+-+-+-+| F| nri| Type |+---------------+F:1 a bit.Forbidden_zero_bit. This one must be 0, as stipulated in the H.Nri:2 a bit.NAL_REF_IDC. Taking 00~11, it seems to indicate the importance of this nalu, such as 00 of the Nalu decoder can discard it without affecting the playback of the image.Type:5 a bit.Nal_unit_type. The type of this NALU unit is su
In the RTP protocol, the source of the ssrc,synchronization source is defined as the RTP packet stream, and the ssrc identifier of the 32-bit value in the RTP header is identified so that it does not depend on the network address. Usually the change of microphone, audio interface, camera, video interface will lead to SSRC changes.In Opal and OpenH323, when the ss
RTP (Real-time Transport Protocol) defines the standard packet format for sending videos and audios Based on the IP network. RTP and RTCP (RTP Control Protocol) work together. RTP carries media streams, while RTCP is used to monitor transmission statistics and Quality of Service (QoS) and assist in synchronization of m
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
Hostzhu comment: mplayer's support for streaming media allows you to use Linux to view live webcast. The promotion of multimedia applications in Linux is not measurable.RTSP/RTP streaming support for mplayerThe Open Source "mplayer" Media Player can now receive and play standards-compLiant RTP audio/video streams, using the "live555 Streaming Media" source codeLibraries.* For example, mplayer can be used to
The Code
In this lab you will implement a streaming video server and client that communicate using the Real-Time Streaming Protocol (RTSP) and send data using the Real-Time Transfer Protocol (RTP ). your task is to implement the RTSP protocol in the client and implement the RTP packetization in the server.
We will provide you code that implements the RTSP protocol in the server, the
This document describes a suitable for bundling, MPEG-2 encoding, RTP protocol can be applied to the video and audio frequency
The payload type of the data. This is the second version. For this type of payload, when it is used in a VoD application system,
Bundling has obvious advantages. This advantage is important enough to sacrifice the modularization of the separated audio video streams.
This type of payload may be used.
1. Introduction
This docume
JMF can be implemented in the RTP media stream playback (playback) and transmission (transmission), mainly by JAVAX.MEDIA.RTP, Javax.media.rtp.event, and the API defined in the JAVAX.MEDIA.RTP.RTCP package is complete. JMF can support specific RTP formats and dynamic loads through a standard JMF plug-in mechanism.
You can play the RTP data stream locally or stor
Transferred from: http://blog.csdn.net/jasonhwang/article/details/7316128The RTP timestamp is calculated at clock rate to represent the time.RTP timestamp represents the time per frame, since one frame (such as I-frame) may be divided into multiple RTP packets, so that multiple RTP timestamp of the same frame are equal. (The frame can be distinguished by the last
H264 RTP Header Parsing ProcessCombined with naldecoder. c Analysis
Protocol Analysis: Each RTP datagram consists of the header and payload. The meaning of the first 12 bytes of the header is fixed, the load can be audio or video data.
The parameter set of the active sequence remains unchanged in the same encoding video sequence, and the parameter set of the active image remains unchanged in the same e
Http://bbs.chinavideo.org/viewthread.php? Tid = 7575
I believe many of you want to play h264 video streaming media like me. However, a newbie often does not know where to start. Using Baidu, Google, and other search materials is a treasure. After N weeks of thinking, I made some achievements. It took a lot of useless effort. I spent a week watching the English protocol, and later I learned that there was a Chinese version, in addition, the goal I have achieved is very simple, as long a
Why does a host inside Nat have access to a Web server outside of NAT, but cannot get RTSP stream Media server stream? Reason: For protocols such as HTTP, the client establishes a socket connection with the Web server, which is monitored by a Web server that binds a fixed TCP port on this port. Clients located behind the NAT randomly select a TCP port connect (2) WEB SERVER. For RTSP streaming media servers, the use of RTP packaging multimedia load, t
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