marketo rtp

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Jrtp Study Notes

RtpsessionFor most RTP applications, the rtpsession class may be the only class used by jrtplib. It can fully process RTCP data packets, so users can focus on real data transmission and receiving.It is important to know that the rtpsession class is not safe under multiple threads. Therefore, you must use some lock synchronization mechanisms to ensure that the same rtpsession instance is not called in different threads.The rtpsession class has the foll

Structure and principle of streaming media client

each frame of data, and play the data on the device after synchronization. From the functional level, the player main module can be divided into four layers: RTSP session control layer, RTP data transmission layer, decoding layer, and display playback layer (1 ). The communication between the player and the server is mainly implemented by the RTSP protocol at the application layer and the RTP protocol (Rea

Ortp Usage Details 2

5. receive and send data 1. Sending process: The rtp_session_send_with_ts interface is called when an application sends data. The parameter is a session handle, data buffer address, data length, and the current timestamp of the application. In this interface, the rtp_session_create_packet interface is called to construct a new message block based on the buffer address and Data Length, and initialize the RTP Header information according to the sessio

Android IOS WebRTC Audio and Video development Summary (57)--a QoS scheme on network transmission

, the impact of packet loss, jitter and chaos on QoS is the most significant, so the QoS solution described below addresses the poor impact of packet loss, jitter, and chaos on the quality of service, as follows:1, the transmission end principle:For real-time audio and video communication, the UDP protocol is used to transmit multimedia data, and the following is a UDP-based RTP protocol for transmitting audio and video data. For different formats of

Jrtplib Study Catalogue and summary

RTP (real-time transport protocol) is generally less of a separate use, and it typically runs with other protocols. For example, RTSP (real-time Streaming protocol) it contains both the RTP,RTCP and the RTSP protocol inside. However, in some systems with very few system resources, the independent use of RTP protocol becomes more and more important. The

Jrtplib Study Catalogue and summary

RTP (real-time transport protocol) is generally less individual use, it will normally run with other protocols. For example, the RTSP (real-time streaming transport protocol) contains both the RTP,RTCP and RTSP protocols. However, in some systems with very few resources, the independent use of RTP is more important. The RTP

Jrtp Programming Model

1) initialization Before using jrtplib for Real-Time Streaming Media data transmission, you should first generate an rtpsession instance to represent the RTP session, and then call the CREATE () method to initialize it. The create () method of the rtpsession class has only one parameter to specify the port number used for this RTP session. 2) Data Transmission After the

Embedded Wireless Video Monitoring System Based on H.264

the command and sends the control command to the monitoring terminal through a wireless network. After the monitoring terminal receives the monitoring task from the central server, it uses the image acquisition module to take on-site images, and will pass through H. 264 the compressed image data is sent back to the central server through the code division multiple access module according to the RTP communication protocol. For wireless users, the cent

Android IOS WebRTC Audio and Video Development Summary (87)--analysis of the implementation of packet loss retransmission Nack in WebRTC

WEBRTC source code, the transmission and reception of video packets is taken as an example, and the implementation of Anck packet retransmission mechanism is deeply analyzed. The main contents include: SDP negotiation Nack, receiving end packet loss determination, NACK message construction, sending, receiving and parsing, RTP packet retransmission. The following are discussed in detail.I. SDP negotiation NACKThe nack is used as the

LIVE555 Source Code Analysis

purposes, you can use this library users can develop traditional running and console applications. By using the subclasses of the custom "Usageenvironment" and "TaskScheduler" abstract classes, these applications can run in a specific environment and do not require much modification. It should be noted that under the graphical Environment (GUI Toolkit), subclasses of abstract class TaskScheduler should be integrated with their own event processing framework when implementing Doeventloop (). Bas

LIVE555 Source Code Analysis

library users can develop the traditional operation and console applications. By using subclasses of the custom "Usageenvironment" and "TaskScheduler" abstract classes, these applications can run in a specific environment and do not require too much modification. It should be noted that under the graphical Environment (GUI Toolkit), subclasses of the abstract class TaskScheduler should be integrated with the event handling framework of the graphical environment when implementing Doeventloop ().

JRTPLIB 3.5.2 Chinese version-part1

This is the latest manual provided on the Joris homepage, although the library has been to version 3.9.0 but the manual function 3.5.2, you can download the English manual from the Jori homepage: http://research.edm.uhasselt.be /~ Jori/page/index. php? N = CS. Jrtplib] JRTPLIB 3.5.2 By: Jori Liesenborgs Jori@lumumba.uhasselt.be

Live555 source code analysis-setup command processing process

Document directory Setup command Overview 2. Transport Header (1.1) 3. Get parameters from subsession (1.2) The setup command provides an overview of the setup command, which is mainly used to negotiate the communication details between the client and the server, such as the communication protocol and address. The most important part of the setup request is the "transport" header. The client needs to send a setup command for each stream in the file. The client can also redirect the

Technical Principles of streaming media

dynamically, the routes selected by each group may be different, so the delay in reaching the client may not be the same, even the first data group may arrive later. Therefore, the cache system is used to compensate for the impact of latency and jitter, and ensure that the Group order is correct, so that the media data can be output continuously without pausing the video due to temporary network congestion. Suitable transmission protocols are required for streaming media transmission. Because

Introduction to HTTP adaptive streaming technology and Its Prospects

delivery technologies mainly fall into two categories: connection-oriented streaming media technologies represented by RTSP/RTP (Real Time Streaming Protocol/Real Time Transfer Protocol, another type is the connectionless HTTP progressive download of mainstream video websites. 1. RTSP/RTP Streaming Media Solution RTSP is a traditional streaming media control protocol. Its Status indicates that the server

The difference between HTTP protocol/RTSP protocol/RTMP Protocol

generally requires 2-3 channels, command and Data channel separation, HTTP and rtmp generally on a TCP channel to transmit commands and data.RTSP, RTCP, RTP differences1:RTSP Real-Time Streaming protocolAs an application-level protocol, RTSP provides an extensible framework that makes it possible to control and on-demand real-time streaming media data. In general, RTSP is a streaming media representation protocol that is primarily used to control dat

The difference between RTSP protocol, RTMP protocol, and HTTP protocol

does not have a specific stream.5:RTSP transmission generally requires 2-3 channels, command and Data channel separation, HTTP and rtmp generally on a TCP channel to transmit commands and data.RTSP, RTCP, RTP differences1:RTSP Real-Time Streaming protocolAs an application-level protocol, RTSP provides an extensible framework that makes it possible to control and on-demand real-time streaming media data. In general, RTSP is a streaming media represent

Page cache processing functions

preemption and assigns some empty radix_tree_node structures to the radix_tree_preloads variable per CPU. The allocation of the radix_tree_node structure is completed by the Slab High-speed cache radix_tree_node_cache. If the structure of the radix_tree_preload () pre-allocated radix_tree_node is unsuccessful, the add_to_page_cache () function terminates and returns the error code-enomem. Otherwise, if radix_tree_preload () is successfully pre-allocated, the add_to_page_cache () function will n

Live555 Study Notes 13-rtpinterface

XIII: rtpinterface details I haven't written a blog for several days. Looking at the source code is really tiring. I need to write my understanding on paper and organize chaotic ideas. This is a headache, so I need passion. But today, passion is coming again. You should have understood the groupsocket class. Theoretically, the classes that need to operate UDP socket should store groupsocket instances. But this is not the case. You can check rtpsink, rtpsource, rtcpinstance, and so on. None of t

Jquery 1.3 and validation plugin 1.5.1

If you want to update the validation plug-in, make sure that you update the validation plug-in to version 1.5.1 at the same time. It is compatible with both version 1.2.6 and 1.3. Major updates include:1. Use the validhandler parameter to replace a form event that must be bound to useless before, the specific demo can access marketo (http://jquery.bassistance.de/validate/demo/marketo)2. Add tiny MCE, sample

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