You need to set the timestamp unit (timestamp) and timestamp increment (timestamp increment) when sending data using Jrtplib. Read some articles on the Internet, carefully wanted to think now just figured out the problem. The RFC3550 description of the timestamp is:
A timestamp (timestamp) 32-bit timestamp reflects the sampling time of the first byte in the RTP packet. (The sampling clock must originate from a timely, monotonically, linearly increme
I wrote an article earlierArticleAnalysis of the format of using RTP for h264 packets: RTP encapsulation of h264. However, it seems that the split and some situations that need attention are not clearly stated, so here we will make a supplement and also serve as our own memo (I don't seem to have a good memory ).
note that the sampling rate of h264 is 90000Hz, so the unit of the timestamp is 1 (s
RTCP RTP protocol format analysis 6: RTCP Sender reportThe sender report consists of three parts, and the fourth part may be extended. Part 1: Header, 8 bytes long, version: 2 bits, RTP version identifier. This version in the RTCP package has the same meaning as that in the RTP package, generally 2 p: fill bit, 1 bit. If set, there are several fill bits at the e
Q: Why are subcontracting sent?
The reasons for the solution and network bandwidth
Sub-code, the situation is as follows:
else if (n->len>1500) {///Gets the Nalu required to send an int k=0,l=0 with a length of 1400 bytes of RTP packets; k=n->len/1400;//requires k 1400 bytes of RTP packet l=n->len%1400;//the last RTP packet needs to load the number of
original articles, Forbidden reprint. otherwise pursued.
The information parsing of RTP header in WebRTC has been explained before.
Here to explain the WEBRTC in the RTP parsing, here is the main explanation of h264 analysis;
About class implementations and related test files that are relevant in VP8 and VP9,WEBRTC;
Regarding the RTP file parsing of H264, t
Real-time audio and video domain UDP is the king
In the Internet, audio and video real-time interaction using the Transport Layer Scheme has TCP (such as: RTMP) and UDP (such as: RTP) two kinds. The TCP protocol can provide a relatively reliable guarantee for data transmission between two endpoints, which is achieved through a handshake mechanism. When the data is passed to the receiver, the receiver checks the correctness of the data. The sender can
The raw stream data obtained from h264 is. Generally, the bitstream structure is SPS, PPS, I frame, P frame ...... SPS, PPS, I frame, P frame ............ When we use RTP to package h264 data, SPS and PPS can directly send I and P frames without sending them. It also depends on the size of I frame and P frame. If it is smaller than MTU, it can be sent directly with the RTP package. If it is larger than MTU,
Payload Structure:
+ ——————— + ———————-+ ————————————
| Payload Header | Table of Contents | Speech data ....
+ ——————— + ———————-+ ————————————-
Payload Header:
0 1 2 3
+–+–+–+–+
| CMR |
+–+–+–+–+
CMR: (4 bits)
Don't know what to do with ~ ~
Can be replaced with the Nal_unit_type in the NAL head completely
Table of Contents:
0 1 2 3 4 5
+–+–+–+–+–+–+
| F | FT | Q |
+–+–+–+–+–+–+
F: (1 bit)
If the frame is the last frame of this RTP packet, then
The Wgscd-picked Rtp/rtcp (real-time transport protocol/real-time Transport Control Protocol) is based on UDP-derived protocols and adds control over real-time transmission. Commonly used for online transmission of real-time video data, such as remote video surveillance, video-on-demand. There is a book called "Multimedia Network Transmission Protocol" on the structure and principle of the 2 agreements to do a more detailed introduction, as if it wa
In the RTP protocol, the source of the ssrc,synchronization source is defined as the RTP packet stream, and the ssrc identifier of the 32-bit value in the RTP header is identified so that it does not depend on the network address. Usually the change of microphone, audio interface, camera, video interface will lead to SSRC changes.In Opal and OpenH323, when the ss
RTCP RTP protocol format Analysis 7: RTCP receiver reportRTCP RTP protocol format analysis 6: RTCP Sender report http://www.bkjia.com/net/201311/255254.htmlThe receiver report and the sender report are basically the same, but the package type is constant 201, and there are no five words of the sender information. The remaining area has the same meaning as the SR package.If no sender or receiver is reporte
We often run a concurrentrequestRTP: ReceivingTransactionProcessor on the rcv side, which is mainly used to process data in RCV_TRANSACTIONS_INTERFACE. this concurrentprogram contains many files. More important: identRVCTPRVCTP: $ Header: rvctp. oc120.0.120
We often run a concurrent request RTP: Processing ing Transaction Processor on the RCV side, which is mainly used to process data in RCV_TRANSACTIONS_INTERFACE. this concurrent program contains man
Note that the x:class in XAML is not changed, and the following 2 red parts are consistent.Namespace RTP. Toolkits{Interaction Logic for Cablelosscalwin.xamlpublic partial class Cablelosscalwin:window{Public Cablelosscalwin (){InitializeComponent ();}}}RTP. Toolkits. Cablelosscalwin "Xmlns= "Http://schemas.microsoft.com/winfx/2006/xaml/presentation"xmlns:x= "Http://schemas.microsoft.com/winfx/2006/xaml"Titl
.
If the Android power supply is a USB host, use Usbdevice. If the peripheral acts as a USB host, use Usbaccessory. Most of the input device mouse and joystick, camera, hubs, etc. belong to the former, namely Usbdevice.
The latter, usually USB devices as the main controller, providing power, communication with the Android device, that is, usbaccessory.
In addition, to handle the mouse, wheel and trackball input, add two new motion event actions:
1.action_scroll, which describes the posit
JMF can be implemented in the RTP media stream playback (playback) and transmission (transmission), mainly by JAVAX.MEDIA.RTP, Javax.media.rtp.event, and the API defined in the JAVAX.MEDIA.RTP.RTCP package is complete. JMF can support specific RTP formats and dynamic loads through a standard JMF plug-in mechanism.
You can play the RTP data stream locally or stor
Twelve h264 RTP packet Timestamp
Let's take h264 as an example.
Void hsf-videortpsink: dospecialframehandling (unsigned/* fragmentationoffset */, The function first checks whether it is the last packet of a frame. If yes, it marks the 'M' and then sets the timestamp. Where does this timestamp come from? It depends on who calls the function dospecialframehandling (). After searching, it is called by multiframedrtpsink: aftergettingframe1. the paramete
About timestamp issues in RTPTimestamp unit: The unit in which the timestamp is calculated is not a unit of seconds, but a unit that is replaced by the sampling frequency, so that the purpose is to be more precise in the timestamp unit. For example, if an audio sample frequency is 8000HZ, we can set the timestamp unit to 1/8000.Timestamp increment: The time difference (in timestamp units) between adjacent two RTP packets.How do I set the increment bet
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 system: Windows 7Software: CMake 3.2.3 + Vis
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 96boolcheckerror (intrtperr); classCRTPSender: Publicrtpses
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