cisco rtp

Network Abstraction Layer Unit type (NALU):The Nalu header consists of a byte with the following syntax:+---------------+|0|1|2|3|4|5|6|7|+-+-+-+-+-+-+-+-+| F| nri| Type |+---------------+F:1 a bit.Forbidden_zero_bit. This one must be 0, as stipulated in the H.Nri:2 a bit.NAL_REF_IDC. Taking 00~11, it seems to indicate the importance of this nalu, such as 00 of the Nalu decoder can discard it without affecting the playback of the image.Type:5 a bit.Nal_unit_type. The type of this NALU unit is su

RTP (Real-time Transport Protocol) defines the standard packet format for sending videos and audios Based on the IP network. RTP and RTCP (RTP Control Protocol) work together. RTP carries media streams, while RTCP is used to monitor transmission statistics and Quality of Service (QoS) and assist in synchronization of m

The Code In this lab you will implement a streaming video server and client that communicate using the Real-Time Streaming Protocol (RTSP) and send data using the Real-Time Transfer Protocol (RTP ). your task is to implement the RTSP protocol in the client and implement the RTP packetization in the server. We will provide you code that implements the RTSP protocol in the server, the

This document describes a suitable for bundling, MPEG-2 encoding, RTP protocol can be applied to the video and audio frequency The payload type of the data. This is the second version. For this type of payload, when it is used in a VoD application system, Bundling has obvious advantages. This advantage is important enough to sacrifice the modularization of the separated audio video streams. This type of payload may be used. 1. Introduction This docume

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

H264 RTP Header Parsing ProcessCombined with naldecoder. c Analysis Protocol Analysis: Each RTP datagram consists of the header and payload. The meaning of the first 12 bytes of the header is fixed, the load can be audio or video data. The parameter set of the active sequence remains unchanged in the same encoding video sequence, and the parameter set of the active image remains unchanged in the same e

Http://bbs.chinavideo.org/viewthread.php? Tid = 7575 I believe many of you want to play h264 video streaming media like me. However, a newbie often does not know where to start. Using Baidu, Google, and other search materials is a treasure. After N weeks of thinking, I made some achievements. It took a lot of useless effort. I spent a week watching the English protocol, and later I learned that there was a Chinese version, in addition, the goal I have achieved is very simple, as long a

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

RTP Header 12 Fixed byte mechanism diagram:0 1 2 30 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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| v=2| p|X| CC |m|PT | Sequence Number |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| Timestamp |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| Synchronization source (SSRC) identifier |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=

Why does a host inside Nat have access to a Web server outside of NAT, but cannot get RTSP stream Media server stream? Reason: For protocols such as HTTP, the client establishes a socket connection with the Web server, which is monitored by a Web server that binds a fixed TCP port on this port. Clients located behind the NAT randomly select a TCP port connect (2) WEB SERVER. For RTSP streaming media servers, the use of RTP packaging multimedia load, t

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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-

Above is the Internet multimedia architecture, we first have a overall impression. RTP (Real-time transport Protocol): RTP provides end-to-end transport for real-time applications, but does not provide any assurance of quality of service. Multimedia data block after compression code processing, first to the RTP packaging into

http://blog.csdn.net/span76/article/details/12913307Offline media only uses the HTTP protocol to read server-side files, and for live broadcast how to achieve, here will use the RTP/RTCP protocolRtp/rtcpRTP is based on UDP protocol, UDP does not have to establish a connection, more efficient, but allow packet loss, which requires more work when re-assembling the mediaRTP is only the parcel content information, and RTCP is the exchange of control infor

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

Https://github.com/whtang/GoRTPGORTP implements some important functions that modify the head and contents of the RTP packet. Most, however, handle only the payload and timestamp of the RTP packet.RTP Data groupingThe packet module implements a buffering mechanism that allows for missed calls, which reduces the dynamic allocation request for memory. Although not necessary, it is recommended that freepacket

This example demonstrates how to use the jrtplib library to encapsulate the RTP protocol in Linux. This routine can be used as a basic routine for streaming media transmission. Only the source code is provided here (these can be found in the official jrtplib files) Sender: /** Sending Program (for Windows and Linux)* For IPv4-based transmission routines, a port number and destination address must be provided.* Reference: http://blog.csdn.net/ipromiseu

Some of the things that have been related to the low-latency transmission of the screen in recent time. Originally wanted to use GStreamer to verify that the RTP over UDP transfer h264 NAL data related, the results found that can not use Playbin to play RTP data! Admittedly, this also has its cause because RTP needs some out-of-band data, which is not simply pass