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
//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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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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