rtp 3000

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Application of Linux-video streaming 4

This article focuses on the frame capture method and transfers the focus from video4linux to the network. In terms of instant transfer of network images, RTP is also the standard used by major manufacturers. In this phase, we will be able to learn how to use jrtplib To add network functions. Video4linux frame capture method In the last issue of xawtv, we saw the image capture function of xawtv, Among them, the most important part is to use video4linux

LIVE555 Analysis: Rtpreceptionstats::noteincomingpacket function

//The Ben function is used to calculate jitter, and this function is called to calculate jitter each time a RTP packet is received. voidRtpreceptionstats::noteincomingpacket (u_int16_t seqNum, u_int32_t rtptimestamp, unsigned Timestampfrequen CY, Boolean Useforjittercalculation,structtimevalResultpresentationtime, Booleanresulthasbeensyncedusingrtcp, unsigned packetsize) { if(!fhaveseeninitialsequencenumber) Initseqnum (seqNum); ++fnumpacketsreceived

Ortp usage 1

I. About ortp Ortp is an open-source software that implements the RTP and RTCP protocols. Currently, the software that uses the ortp library is mainly Linphone (a software for video and voice calls based on IP addresses ). As the RTP Library of Linphone, ortp guarantees the transmission of voice and video data based on the RTP protocol. Ii. Source Code constru

How to Use mplayer

RTP;9. Play a DVDMplayer DVD: // 10. Specify the subtitle file. Mplayer-sub 11. Use Subtitles with language-specific code Mplayer DVD: // 12. Synchronization Solution Repair efforts Mplayer-autosync 30-MC 2.0 Not repaired Mplayer-autosync 0-MC 0 13. Playing on slow CPUMplayer-frameddrop 14. playlistMplayer-playlist 15. mpalyer specifies the cache Mplayer-Cache 8192-playlist Mplayer-Cache 8192-Cache-min 50-playlist Mplayer/tmp/media-Cache

[Zz] RTCP

RTCP] RTP (Real-timetransportprotocol) is a transmission protocol for multimedia data streams on the Internet. RTP is defined to work during one-to-one or one-to-many transmission. It aims to provide time information and implement stream synchronization. RTP usually uses UDP to transmit data, but RTP can also

SIPP User Guide)

verification. 3) pcap play is supported, but password verification is not supported: (pcap play means RTP speech can be performed, but no 407 auth verification is performed) A) # tar-xvf sipp-1.1rc6.tar.gzb) # Cd sipp-1.1.rc6c) # Make pcapplay 4) Support for pcap audio file playback and password verification: (407 auth verification supported) A. # tar-xvf sipp-1.1rc6.tar B) # cd sipp-1.1.rc6 C) # Make pcapplay_ossl1.2 SIPP usage When using SIPP for

VoIP bookmarks from Klaus Darilion

Document directory RTP Stacks (mainly open source C/C ++ stacks) SIP Stacks RTP Applications SIP Phones (SIP User Agents) SIP Test Utility SIP Applications (Proxy, Location Server) Sip Express Router (ser) Ser Media Server (sems) STUN server and clients NAT traversal ALG (application level gateway) Below you will find descriptions and links to SIP and RTP

Key IP-based H.264 Technology

I. IntroductionH.264 is the latest video coding standard for ITU-T, known as ISO/IEC14496-10 or MPEG-4 AVC, and is a new product jointly developed by the video coding Expert Group of Motion Image Expert Group (mPEG) and ITU. H.264 is divided into two layers, including the video encoding layer and network adaptation layer. The video encoding layer processes block, Macro, and chip data and tries its best to be independent from the network layer. This is the core of video encoding, including many t

[Reprint] video encoding (h264 overview)

consists of a 1-byte header, three fixed-length fields, and an uncertain number of encoding segments. Header mark Syntax: NALU type (5bit), importance indication bit (2bit), and prohibition bit (1bit ). NALU type: 1 ~ 12 used by H.264, 24 ~ 31 is used by applications other than H.264. Importance indication: indicates the importance of the nal unit for reconstruction. The greater the value, the more important it is. Bit prohibited: when the network discovers that the nal unit has a bit error, yo

"rfc3605" Real Time Control Protocol (RTCP) attribute in session Description Protoco

numbers are not enough. The SIP messages use the "encoding defined in SDP" [RFC2327] to describe the IP addresses and TCP or UDP ports used by the Various media. Audio and video are typically sent using RTP [RFC3550], which requires two UDP ports, one for the media and one for the C Ontrol Protocol (RTCP). SDP carries only one port number per media, and Huitema standards Track [Page 1] RFC 3605 RTCP attribute in SDP Oct

Comparison between MPEG2 TS and Isma

a single first-class transmission after the video and audio streams are reused, the support of MPEG-2 ts over IP for multi-track, multi-word screen and future interactive scenarios needs to encapsulate all stream information in a unified manner, this is a waste of bandwidth that users do not need. If video communication, VoIP and other traffic business, MPEG-2 ts over IP Mode also need to expand to support. 2. Support for Business Performance Isma introduces the

Key technologies and applications of H.264 Based on IP Networks

H.264 is the latest video coding standard for ITU-T, known as ISO/IEC14496-10 or MPEG-4 AVC, and is a new product jointly developed by the video coding Expert Group of Motion Image Expert Group (mPEG) and ITU. H.264 is divided into two layers, including the video encoding layer and network adaptation layer. The video encoding layer processes block, Macro Block, and chip data and tries its best to be independent from the network layer. 264 is the latest ITU-T video coding standard, known as ISO/I

Getting started with ortp Library

I. Introduction Ortp is a database that supports RTP and rfc3550 protocols. It has the following features:(1) It can be written in C language and can work on Windows, Linux, and UNIX platforms.(2) implements the rfc3550 protocol and provides easy-to-use APIs. Multiple configurations are supported. rfc3551 is the default configuration.(3) supports multiple RTP sessions in a single thread and adaptive jitter

A convenient way to process server project logs

It is found that project log processing by O M colleagues is convenient and worth learning. Even in the development environment, this process is good.For example, the project RTP is deployed under the/home/www/RTP directory.Create the/usr/userfile/logs directory to store logs.Create the/home/www/defonds-config/RTP directory to store configuration files (such as

Asterisk 1.8 SIP protocol stack Analysis 2

dialog_initialize_rtp function to initialize the RTP information of the peer. Whether the peer has RTP. If yes, the encoding is set. When setting the RTP engine, it must be noted that the RTP protocol stack was greatly changed at the beginning of asterisk1.8. By default, the RTP

Jrtplib packet receiving process

Recently, RTP has naturally involved the jrtplib library, reading code 3.7.1, spare time, and some excerpt. I hope you can remember it quickly in the future. I also hope your friends can read it and point out improper information, we provide you with valuable suggestions to learn and make progress together.Next, the source code analysis notes are based on: # ifndef rtp_support_thread. The author uses related classes for background threads to process t

Source Code address of the VoIP open-source project

VoIP bookmarks from Klaus darilion Below you will find descriptions and links to sip and RTP stacks, applications, test utilities, SIP proxies, SIP pbxs and stun server and clients. most of them are open source :-), but not all of them If you have any comments please feel free to contact me: --> Klaus. darilion at pernau. at There are also other VoIP related portals and link collections. Note: I mainly searched for C/C ++ stacks and applications. the

Introduction to the WEBRTC audio processing process

This article provides an overview of the WEBRTC audio processing flow, as shown in the following diagram: WebRTC an audio session is abstracted into a channel, such as A and b for audio calls, a needs to establish a channel and b for audio data transmission. The figure above has three channel, each channel contains codec and real-time Transport protocol (real-time Transport Protocol,RTP)/ The real-time Transport Control Protocol (real-time control PR

Detailed explanation of H3C router configuration commands

socket display ip statistics display acl all acl number acl-number match-order auto/config acl-number (2000- 2999 is the basic acl 3000-3999 is the number reserved by the advanced acl for the Administrator) rule deny/permit protocal Access Control [h3c] acl number 3000 [h3c-acl-adv-3000] rule permit tcp source 129.0.0 0.0.255.255 destination 202.38.160.0 0.0.255

H264 frame structure Analysis and frame judging

encounter two bytes consecutive 0, insert a byte of 0x03. 0X03 is removed when decoding. Also known as shelling operations.Encoding process:1. Packaging the Sodb of the VCL output into Nal_unit,nalu is a general encapsulation format, which can be applied to the sequential byte stream and the IP packet switching mode.2. For different transport networks (circuit-switched | packet switching), the Nal_unit is packaged into a package format for different networks (such as encapsulating Nalu into

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