rtp 3000

:10.0.0.12:5060to:128.97.41.56:5060 (address of the next hop)SDP Body IP address for RTP:10.0.0.12:100025.NAT server modifies the IP address of its package. Sent to sip ALG.6.SIP ALG accepts this and finds that its packet has a different IP address and SIP IP address, so it is judged to be NAT, so it modifies its associated SIP IP address.and check whether its body contains SDP information, if it is, and there is a

. Check whether the body contains SDP information. If yes, and there is an RTP address, the sip alg will request a public RTP address from the RTP proxy to replace the original RTP address. IP packet IP Address: From: 128.97.41.56: 5060 T 128.96.63.25: 5566 Sip msg ip Address: From: 128.96.41.1: 5678 T 128.96.63.25: 55

:128.96.41.1:5678 to:128.97.41.56:5060 (SIP ALG) SIP MSG IP Address: from:192.168.1.10:5060 to:128.97.41.56:5060 SDP Body IP address for RTP: 192.168.1.10:10024 3. SIP ALG accepts the invite and finds that its packet has a different IP address and SIP IP address, so it is determined to be NAT, so it modifies its associated SIP IP address. and check whether its body contains SDP information, if it is, and there is a

external interface, it is managed by the Voebase, and the corresponding channel is selected by the index number.Voebase*base = Voebase::getinterface (pvoeengine);int Ch0 =base->createchannel ();2) Network Port settingsThe audio is sent out via RTP and RTCP, RTP and RTCP are implemented using UDP, and network ports and addresses need to be configured.Set up 3000

layer or storage media, and provides early information to provide video encoding and external world interfaces. NALU: defines basic formats that can be used for group-based and bit stream-based systems. RTP encapsulation: only for the local nal Interface Based on the nal unit. Three different data forms: Sodb data Bit String --> the original encoding data Rbsp original byte sequence load --> after sodb, add the ending bit (rbsp trailing bits is a bi

Real-time transfer implementation based on the jrtplib library in Linux I. RTP is a standard protocol and Key Technology for Real-Time Streaming Media transmission. Real-Time Transport Protocol (PRT) is a network protocol used to process multimedia data streams over the Internet. It can be used in one-to-one (unicast, unicast) scenarios) or you can transmit streaming media data in real time in a one-to-multiple (Multi-play) network environment.

Prerequisite Description:　　When doing the GStreamer project, it is necessary to actively send the data collected from the device to the server.This allows you to proactively send data to the specified server using the Tcpclientsink and Udpsink plugins.Tcpclientsink usageNote: If you want to proactively send data to the server, you can transfer it via the Tcpclientsink plugin.The specific code is as followsData-client:Send to Linux:0 - - - 5 4000000 640,480 -O-| gst-launch-1.0 fdsrc! h264pars

Use DirectShow to implement QQ's audio/video chat function Currently, popular instant messaging tools, such as MSN and QQ, all implement the video and audio functions. Through video and audio, we can better communicate with our friends through the network, this article uses DirectShow technology to simulate QQ to achieve video and audio acquisition, transmission, and basically implement the QQ video and audio chat function. The main function of the network video/audio system is the collection of

) protocol is a connection-oriented transport protocol, the communication needs to establish a connection, transmission delay is large, TCP recognition and retransmission mechanism, flow control mechanism can ensure reliable data transmission, but the processing process is complex and inefficient, for audio and video streaming , frequent acknowledgement and retransmission cannot guarantee the real-time transmission of data, so it is relatively unsuitable for the transmission of video images. Su

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 (NAL) is very suitable for real-time transmission of

016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very much. Therefore, it is very important. In general

Turn from: http://blog.csdn.net/lixiaowei16/article/details/53407010 Audio and video synchronization is related to the most intuitive user experience of multimedia products, audio and video media data transmission and rendering playback of the most basic quality assurance. If the audio and video is not synchronized, it may cause delays, such as cotton, etc. very affect the user experience phenomenon. Therefore, it is very important. Generally speaking, the audio and video synchronization maint

2016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very much. Therefore, it is very important. In gener

Section 5. transmitting and receiving media JMF and real-time transmission protocol (RTP) Many friendly network features are directly built in JMF, which makes it easy for client programs to transmit and receive media over the network. When a user on a network wants to receive media streams of any type, it does not need to wait for all broadcasts to be downloaded to the machine before watching the media; users can watch broadcasts in real time. This c

Transport Protocol III, RTSP (real Time streaming Protocol) The above streaming media playback based on progressive downloading can only support on-demand and cannot support live broadcast, the rate at which the media stream data arrives at the client is not precisely controlled, and the client still needs to maintain a buffer storage space of the same size as the media file on the server, waiting for a long buffer time before it can begin playback, resulting in poor real-time performance, Duri

Implementation of real-time transmission based on Jrtplib library under LinuxRTP is a standard protocol and key technology for real-time streaming media transmission.Real-time transport protocol (real-time transport PROTOCOL,PRT) is a network protocol for processing multimedia data streams on the Internet, which can be used in one-to-one (unicast, unicast) or one-to-many (multicast, multicast), the real-time transmission of Liu Media data is realized in the network environment.

Real-time transmission protocol RTP 1. RTP protocol:RTP (Real-Time Transport Protocol) was originally used in 1970s to try to transfer audio files, divided into several parts to transmit voice, time signs and queue numbers. After a series of developments, the first version of RTP was released by a laboratory in the United States in August 1991. By the 1996 s of t

RTP Reference rfc3550/rfc3551 Real-Time Transport Protocol) is a transport layer protocol for multimedia data streams on the Internet. The RTP protocol details the standard packet formats for transmitting audio and video on the Internet. RTP is often used in streaming media systems (with RTCP protocol), video conferencing and one-click push to talk systems (with

RTP Reference Document Rfc3550/rfc3551 Real-Time Transport Protocol) is a transport layer protocol for multimedia data streams on the Internet. The RTP protocol details the standard packet formats for transmitting audio and video on the Internet. RTP is often used in streaming media systems (with RTCP protocol), video conferencing and one-click push to talk syste

transmitting real-time audio and video data. To achieve real-time transmission of audio and video data, we need to seek other channels.4. RTP protocol is suitable for real-time video and audio transmission. RTP (Real-time transport protocol)/RTCP (Real-Time Transport Control Protocol) is an application-oriented transport layer protocol, it does not provide any guarantee of Transmission reliability and tra