rtp transmission

Hostzhu comment: mplayer's support for streaming media allows you to use Linux to view live webcast. The promotion of multimedia applications in Linux is not measurable.RTSP/RTP streaming support for mplayerThe Open Source "mplayer" Media Player can now receive and play standards-compLiant RTP audio/video streams, using the "live555 Streaming Media" source codeLibraries.* For example, mplayer can be used to

I. Jrtplib INTRODUCTIONRTP is the best way to solve the problem of streaming media real-time transmission, and Jrtplib is a C + + language implementation of the RTP library, it is fully compliant with RFC 1889 design, now can run in Windows, Linux, FreeBSD, Solaris, UNIX and VxWorks and many other operating systems. Before you can use Jrtplib, you need to compile it.Two. Platforms and software usedOperating

In the previous article we introduced some basic knowledge of the RTP protocol, below we describe how to use jrtplib this library to transmit H264 encoding.JRTP transmission: OK, here is the example I wrote about sending H264 packets using JRTP, which can be explained in detail. The sending side can also receive RTCP packets sent by the receiving end. #define MAX_RTP_PKT_LENGTH1360#defineH264 96boolcheckerr

FEC (Forward error Correction) forward error correction UDP\RTP used to improve wireless and other network packet loss problems The algorithm is not introduced at this stage. Idea: FEC ENCODE adds redundant packets, and when the wireless network drops packets, the receiver uses redundant packets to decode the lost packets. Example: 10 packets, the code will add 2 packets, a total of 12 packets sent to the receiving end, the receiver lost the 5th an

Transferred from: http://blog.csdn.net/jasonhwang/article/details/7316128The RTP timestamp is calculated at clock rate to represent the time.RTP timestamp represents the time per frame, since one frame (such as I-frame) may be divided into multiple RTP packets, so that multiple RTP timestamp of the same frame are equal. (The frame can be distinguished by the last

Multicast routing is a good technology. It broadcasts data over the Internet. Unlike broadcast, due to broadcast storms, the router prohibits cross-route transmission of broadcast data. Multicast solves this problem. Currently, m$ software such as NetMeeting and WMS are widely used in multicast. Here we will discuss how to build your Linux into a multicast router.(Generally, gateways and routers do not support multicast data packets ). 1.

The RTP timestamp is calculated at clock rate to represent the time. RTP timestamp represents the time per frame, since one frame (such as I-frame) may be divided into multiple RTP packets, so that multiple RTP timestamp of the same frame are equal. (The frame can be distinguished by the last

RTP over RTSP (TCP) (i)solution for mixed delivery of RTP over RTSP packetsRTSP-RTP over TCP-to-use TCP communication, the need to request TCP connection during RTSP SETUP. The RTSP client needs to request a TCP connection in the setup phase for the sent Setup command with which you want to use a TCP connection. The Setup command should include transport in the f

VLC plays the. 264 file sent by RTP package1, to have a server that sends RTP packets of 264 files;The specific code is as follows:Rtp.h#include Rtp.cpp #include "rtp.h" #include 2, download VLC;3, make the following settings:1> setting W.SDPA, open Vlc->media->open network streamb,-->network->rtp://@192.168.1.101:1234C,-->file->add...w.sdp->playHere to talk ab

# Ifndef _ h_rtpheader # DEFINE _ h_rtpheader /* ++V-version. Recognize the RTP version.P-gap (padding ). When this parameter is set, the data packet contains one or more additional gap groups, which do not belong to the payload.X-extended bit. When setting, after the Fixed Header, set an extension header according to the specified format.CSRC count-the number of the CSRC identifier (after the Fixed Header.M-mark. Mark defined by the profile file. Al

Video 1 90000 ITU-T H.261 video RFC 4587 32 MPV Video 1 90000 MPEG-1 and MPEG-2 video RFC 2250 33 Mp2t Audio/Video 1 90000 MPEG-2Transport Stream Video RFC 2250 34 H263 Video 90000 H.263 video, first version (1996) RFC 3551,RFC 2190 Dynamic H263-1998 Video 90000 H.263 video, second version (1998) RFC 3551,RFC 4629,RFC 2190 Dynamic H263-2000 Video 900

1 RTP Message Format The RTP message format is as follows, see RFC1889 (most audio visual coding formats are consistent, some changes are shown in various coded RTP protocols): A. RTP Header format Simple introduction 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-

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

, and sensitive to transmission latency and jitter. However, under certain circumstances, packet loss can be allowed, that is, a certain degree of Transmission Error code is acceptable. In addition, the streaming media service must meet the needs of broadcast and multicast applications, and must have the ability to adjust the video transmission quality according

---------------------------------------------------------------------------------------------------------------------------------- /*Here's a small IPv4 example: it asks for a portbase and a destination andStarts sending packets to that destination.This is a small IPv4 sample program: it requests a starting port and a destination address and then starts sendingSome packages to the target address.*/ # Include "rtpsession. H"# Include "rtppacket. H"# Include "rtpudpv4transmitter. H"# Include "rtp

RFC3984 is the specification of H. Baseline streaming in RTP mode, where only fu-a subcontracting is discussed, as the work is just used, it is written down. Fu_a a kind of fragmented packet, is to encapsulate an oversized Nalu unit into a plurality of RTP packets, which is different from the previous kind of single nalu encapsulated into a single RTP package, of

connected at the modem speed. Moreover, because of the loss of error information is ignored, the network congestion or problems, video quality is poor, and no sequential streaming video quality is good. Live streaming requires a specific server, such as QuickTime streaming server, Realserver, and Windows Media server. These servers allow you to perform more levels of control over media delivery, so system settings and administration are more complex than standard HTTP servers. Live streaming al

Wireshark captures packets to obtain rtp ts Stream Data and saves it as a TS File First parse the RTP stream 2. Click statistics-RTP-show all streams in the menu bar] 3. display the analyzed RTP streams in Wireshark: RTP streams and select the first MPEG-2, whi

These two days have been blocked by this problem. The VxWorks document does not mention how the root task of RTP of VxWorks receives the event from the kernel mode. The root Rask here refers to the task automatically generated after RTP spwan. It is said that some other companies also use this workaround method. Currently, WinDriver is helpless. First, we need to clarify one concept:

RTP supports shipping Steaming of different codec versions. The clock rate varies with codec versions. RTCP is required for synchronization between different media sets. Here is a brief introduction to their mechanism. In each RTCP Sr package, there is an RTP time and an NTP time. It clearly indicates the absolute time corresponding to the RTP time, although the