rtp protocol

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 S

RTCP RTP protocol format Analysis 1: The task of sequential RTCP is to provide feedback on the delivery quality of the data streams transmitted over RTP, such as packet loss rate, jitter, bandwidth, and rate; when it is detected that the quality is poor, it will make some adjustments based on its own settings when the next packet is sent to achieve optimization.

RTCP RTP protocol format analysis 6: RTCP Sender reportThe sender report consists of three parts, and the fourth part may be extended. Part 1: Header, 8 bytes long, version: 2 bits, RTP version identifier. This version in the RTCP package has the same meaning as that in the RTP package, generally 2 p: fill bit, 1 bit.

RTCP RTP protocol format Analysis 7: RTCP receiver reportRTCP RTP protocol format analysis 6: RTCP Sender report http://www.bkjia.com/net/201311/255254.htmlThe receiver report and the sender report are basically the same, but the package type is constant 201, and there are no five words of the sender information. The

#pragma comment(lib, "jrtplib.lib") #pragma comment(lib,"jthread.lib") #pragma comment(lib,"WS2_32.lib")#endif2. Add a header fileStep one: Put all the. h files together, such as myJRTPLIBHeader inside, and then add include .Step Two: Configuration Properties---Properties---add include directories, general--Five. Reference linksHttp://www.cnblogs.com/skyseraph/archive/2012/04/07/2435540.html Copyright NOTICE: This article for Bo Master original article, without Bo Master permission no

3. Dynamic Distribution and implementation of cache areas. Based on this fact, the audio data is much smaller than the video data. Therefore, the audio buffer and the video buffer are managed separately. According to statistics, the RTP data packet size of the same media stream is similar. For example, the audio size is about several hundred bytes, and the video size is 1.3 Kbytes. The video buffer is divided into two parts. Some are in the Active Dat

As the sender of multicast data, adding arm_linux to multicast groups is not required. You can add or not. Arm_linux data sending program: (only functional functions are provided) Void multicast_thread (){Rtpsession sess;Int portbase = 6000;Int status;// Int length;Int loop;Bool iscast;Unsigned long destip;Unsigned long localip;Char * buff = "multicasting test ";Int length = strlen (buff );Destip = inet_addr ("233.0.0.1 ");Localip = inet_addr ("172.29.26.101 ");If (destip = inaddr_none | localip

Original address: http://www.cnblogs.com/imystr/p/4026639.html OSI Seven layer modelLayer-Functional TCP/IP protocol family in OSIApplication-layer file transfer, email, file service, virtual terminal tftp,http,snmp,ftp,smtp,dns,telnetPresentation layer data format, code conversion, data encryption No protocolSession layer release or establish contact with other contacts no agreementThe Transport layer provides end-to-end interface tcp,udp (

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. Thi

summed up in a simple sentence: RTSP initiated/terminated streaming media, RTP transmission stream Media data, RTCP control of RTP, synchronization. The reason for this is a bit confusing, because the CTC standard does not have RTCP requirements, so in the standard RTSP code does not see the relevant part. In the private RTSP code, the control, synchronization, etc., are implemented in the

Introduction in a simple sentence summary: RTSP initiate/End streaming media (application layer), RTP transport Stream Media data (transport layer), RTCP to RTP control, synchronization (transport layer). The reason for this is a little confusing, because the CTC standard does not require RTCP, so in the standard RTSP code does not see the relevant parts. In the private RTSP code, the control, s

. 134/// generate random identifier algorithm see directory A.6. Although the probability of choosing the same identifier is very small, 135 ///But all RTP implementation must detect and resolve the conflict. 136 ///The 8th chapter describes the probability and resolution mechanism of the conflict and the RTP-level detection mechanism, according to the unique ssrcidentifier forward loop. 137///I

1. Introductionwith the development of information industry, people's demand for information resources has been gradually transition from text and pictures to audio and video, and more and more emphasis is placed on the real-time and interactivity of acquiring resources. But people are faced with another unavoidable embarrassment of having to spend a lot of time waiting to transfer files while seeing vivid and clear media demonstrations on the web. In order to solve this contradiction, a new med

-time multimedia transmission. RTP is an application-oriented transport layer protocol that does not provide any guarantee of Transmission reliability and traffic congestion control mechanisms. RTP is located on top of UDP (User datainprotocol. Although UDP is not as reliable as TCP and cannot guarantee the service quality of real-time services, RTCP needs to mon

RTP provides an end-to-end network transmission function, suitable for applications that transmit real-time data (multicast or Unicast Network Services), such as audio and video. RTP does not require resource reservation or guarantee the service quality of real-time services. The Data Transmission Control Protocol RTCP is enhanced. This

) multicast. In a network built using Multicast technology, the router is allowed to copy data packets to multiple channels at a time.(3) On-demand video and broadcast. The VOD connection is an active connection between the client and the server. In the VOD connection, you can select a content item to initialize the client connection. You can start, stop, back, fast forward, or pause a stream. Broadcast means that the user passively receives a stream. During the broadcast process, a separate cop

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. Thi

(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, service

address for data transmission. The RTP protocol allows multiple destination addresses for the same session. You can call the adddestination (), deletedestination (), and cleardestinations () of the rtpsession class () method. For example, the following statement indicates that the RTP session sends data to Port 6000 of the local host: Unsigned long ADDR = ntohl

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