The data in the live555 is sent in the end to be sent using the RTP protocol, which describes the RTP package format.RTP PacketRTP is based on the UDP protocol, and the RTP server passes the UDP protocol, typically sending an RTP packet each time. The client reads the data and then plays it by parsing the
1. Network Abstraction Layer Unit type (NALU)Nalu is H264 used for network transmission of the unit type, a complete Nalu unit is generally 0x000001 or 0x00000001 start, followed by Nalu head and NALU data; When we were transmitting in the network, Will remove the start 0x000001 or 0x00000001, and you will typically need to replace them with the head of the RTP payload (1 bytes);Where NALU data is rbsp data;The Nalu header consists of a byte with the
Here we will mainly analyze the content about RTP packaging and sending in live555. After the play command is processed, the RTP packet is sent. (In fact, a RTP packet will be sent before the response packet of the play command is sent, and the transfer has started here)The RTP packet is sent starting from the call of
(1) Introduction to concepts
RTP (Real-Time Transport Protocol) is defined as a transmission protocol for transmitting real-time data, such as audio, video, and analog data, compared with the traditional high-reliability data transmission transport layer protocol, it focuses more on real-time data transmission, services provided by this Protocol include data sequence numbers, time tags, and transmission control. R
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 Jrtplib and Ortp databases, one is the c ++ inte
terminate a session Process attended by one or more participants. Compared with H.323, SIP is simpler, more flexible, and easier. Has gradually become the focus of attention. currently, many organizations have implemented the SIP protocol stack and made it open-source for developers to use it conveniently and quickly, such as open-source projects such as Osip and resiprocate, these open-source code implements the SIP protocol stack according to the rfc3261 and other sip-related standards. Base
[GO] Streaming Media protocol introduction (RTP/RTCP/RTSP/RTMP/MMS/HLS)http://blog.csdn.net/tttyd/article/details/12032357RTP reference Documentationrfc3550/rfc3551Real-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
RTP reference Documentationrfc3550/rfc3551Real-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-t
http://blog.csdn.net/tttyd/article/details/12032357RTPReference documentsrfc3550/rfc3551Real-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-
RTP Message Format
The RTP message consists of two parts: header and payload. The RTP header format is shown in the following illustration, where:
L The version number of the V:RTP protocol, which accounts for 2 digits, and the current protocol version number is 2.
L P: Fill mark, 1 digits, if p=1, then fill one or more additional eight-bit groups at the en
Warning: The original page of this article http://www.cnblogs.com/moonvan/archive/2011/09/11/2173448.htmlOn the basis of a slight modification, if there is infringement, please notify the deletion, thank you!Streaming Media Protocol
There are two main ways to transfer audio and video information on the Internet today: Download and streaming.
In the case of download, the user needs to download the entire media file locally before the media file can be played.
Streaming refers to
Recently made video codec part, transmission using RTP protocol. Make a record of learning1. Introduction The real-time Transport protocol (real-time Transport Protocol or abbreviated RTP) is a network transport protocol, which is a multimedia transmission by the IETF Working Group 1996 in RFC Published in 1889. RTP is defined as a transport protocol for transmi
in streaming media technology.H.264/AVCIt is a new generation of video 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 layer design of the video encoding layer (VCL) and network extraction layer (
=====================================================Audio-visual data Processing Primer series articles:Getting started with visual audio data processing: RGB, YUV pixel data processingGetting Started with AV data processing: PCM Audio sampling data processingGetting Started with AV data processing: Analysis of video stream in H.Getting Started with AV data processing: AAC audio bitstream parsingGetting Started with AV data processing: FLV Encapsulation Format parsingGetting Started with AV dat
RTP packet: a packet with an RTP header on the start, that is, RTP packet. Generally, in the bottom layer protocol, a packet contains only one RTP packet, but through some special encapsulation methods, the bottom layer packet can also contain more than one RTP packet.
The Header Format of RTP is as follows:
Version Number (V): 2 bits, used to mark the RTP version used.
Fill bit (P): 1 bit. If this position is set, the end of the RTP package contains additional fill bytes.
Extended bit (x): 1 bit. If this location is used, an extended header is followed by a fixed RTP Header.
CSRC co
5.3.1 RTP Header Extension
The following provides an extension mechanism to allow some implementation requirements to experiment with the new load format-independent feature that carries additional information in the RTP data header. This mechanism is designed for other unscalable implementations to ignore these header extensions.
Note that this header extension is intended for some limited purposes. Most o
Session. begindataaccess (); If (session. gotofirstsource () {do {rtppacket * packet; while (packet = session. getnextpacket ())! = 0) {cout
For the establishment of the jrtplib environment, I can refer to my previous summary. Now I will mainly talk about how to learn example under jrtplib3.71. I. sample is a simple IPv4 column that implements RTP data transmission on the local machine. 1. initialization. We know that
Session. begindataaccess (); If (session. gotofirstsource () {do {rtppacket * packet; while (packet = session. getnextpacket ())! = 0) {cout
For the establishment of the jrtplib environment, I can refer to my previous summary. Now I will mainly talk about how to learn example under jrtplib3.71. I. sample is a simple IPv4 column that implements RTP data transmission on the local machine. 1. initialization. We know that
1 Streaming Media protocol
There are two main ways to transfer audio and video information on the Internet today: Download and streaming.
In the case of download, the user needs to download the entire media file locally before the media file can be played. Streaming refers to the pre-processing of multimedia (reducing quality and efficient compression) before transmission, and then using a caching system to ensure that the data is continuously and correctly transmitted. T
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