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
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
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
, the impact of packet loss, jitter and chaos on QoS is the most significant, so the QoS solution described below addresses the poor impact of packet loss, jitter, and chaos on the quality of service, as follows:1, the transmission end principle:For real-time audio and video communication, the UDP protocol is used to transmit multimedia data, and the following is a UDP-based RTP protocol for transmitting au
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
,nal_unit_unspecified_57,NAL_UNIT_UNSPECIFIED_58,nal_unit_unspecified_59,NAL_UNIT_UNSPECIFIED_60,Nal_unit_unspecified_61,NAL_UNIT_UNSPECIFIED_62,NAL_UNIT_UNSPECIFIED_63,Nal_unit_invalid,};Below receives the FU group packs the way, the FU packet header format is as follows:
Fus header contains two bytes of Payloadhdr, a byte of Fu Header,fu header and H264, the structure of the following figure, including the start bit (1b), stop bit (1b), Futype (6b)
PAYLODHDR two of its own assignment, in fact,
The payload type (Payload type) in the RTP packet has a column length of 7 bits, so RTP can support 128 types of payload that are not available. This field is used to indicate the type of encoding used by the sound or image, and is determined by the sending side, but of course the receiver has the ability to handle it. If the sender decides to change the encoding mode in the middle of the session or broadca
We used the RTP protocol in the previous issue and were configured to package the video data, which we reorganized and displayed on the receiving end of the data we sent. Finally, it is extended to support multi-client video sending, and display on the receiving end of the screen. Complete the remote monitoring simulation.Let's start with one.private bool Newrtppacket (Rtppacket packet) {if (! Clients.containskey (packet. SSRC)//If
Wireshark RTP parser DoS Vulnerability (CVE-2014-6421)
Release date:Updated on:
Affected Systems:Wireshark 1.12.0Description:Bugtraq id: 69855CVE (CAN) ID: CVE-2014-6421
Wireshark is the most popular network protocol parser.
Wireshark 1.12.0 has a denial of service vulnerability. Attackers can exploit this vulnerability to crash affected applications.
*>
Suggestion:Vendor patch:
Wireshark---------The vendor has released a patch to fix this secu
As the sender of multicast data, adding arm_linux to multicast groups is not required. You can add or not.
Arm_linux data sending program: (only functional functions are provided)
Void multicast_thread (){Rtpsession sess;Int portbase = 6000;Int status;// Int length;Int loop;Bool iscast;Unsigned long destip;Unsigned long localip;Char * buff = "multicasting test ";Int length = strlen (buff );Destip = inet_addr ("233.0.0.1 ");Localip = inet_addr ("172.29.26.101 ");If (destip = inaddr_none | localip
(1)(2)(3)-------------AUTHOR:PKF------------------time:2015-1-6Later discovered is that the server address of the connection is specified so that it can not play, and then with Wireshark capture a lot of UDP data came out, is not a port problem, and RTSP head problemIf it is an RTSP contract, the packet header will have a 4-byte symbol for a specific frame:Framingheader[0] = ' $ ';FRAMINGHEADER[1] = Streamchannelid;FRAMINGHEADER[2] = (u_int8_t) ((packetsize0xff00) >>8);FRAMINGHEADER[3] = (u_int8
After more than 20 days of sleep in a month (I really didn't take a nap), I finally completed the RTP/RTSP forwarding server (or proxy server )!!
New servermediasubsession, Demux, source, and sink classes are added based on the live555 architecture. (many classes use existing live555 classes as much as possible ). both VOD and real-time streams can be forwarded.
When Forwarding Real-time streams, we make full use of the existing data stream structur
H264.h Header File Contents:
#include
RTPSend.cpp File Contents:
#include
Save the following text as a RTPPARA.SDP file
M=video 1234 RTP/AVP 96a=rtpmap:96 H264A=framerate:30C=in IP4 192.168.0.25
Open the file with VLC and run the compiled send program.
Note that the IP of the header file and the SDP file should match the IP of the experimental machine.
The code modifies a memory bug based on the Naldecoder program. Thanks to the predecesso
that TCP is not suitable for real-time video/audio transmission is that its packet header is larger than that of UDP. The TCP packet header is 40 bytes, while the UDP packet header is only 12 bytes. In addition, these reliable transport layer protocols cannot provide time stamps and encoding and decoding information, which is precisely the application of the receiver (client ). Program Required. Therefore, TCP is not suitable for real-time
RTP communication between PC and ARM-based JRTPLIB-general Linux technology-Linux technology and application information. For details, refer to the following section. In a cross-compiled library on linux, copy the library file to/usr/lib under nfs, compile it in WINDOWS, and test it in windows, however, communication between ARM and windows fails. I found some information on the Internet, saying that the PC side has a small byte order and the ARM side
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