rtp protocol

|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| Synchronization source (SSRC) identifier |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+| Contributing Source (CSRC) identifiers || .... |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+Load type Payload type (PT): 7bitsThe RFC defines this payload type for some of the earlier formats. But later, such as H264 is not allocated, then use 96来 instead. So now more than 96 does not represent a spe

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 pay

) 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 tran

The asterisk kernel (hereinafter referred to as the kernel) provides a series of RTP-related API functions. When using different RTP stacks, these Apis provide a unified access method for the RTP module. After these APIs are encapsulated, any module using RTP will not feel the difference between the underlying stack. F

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

the following steps, in order:1. Download and build the "live555 Streaming Media" libraries. You may wishTo move the resulting "live/" directory to "/usr/local/", "/usr/local/lib /",Or "/usr/lib/". (Note: if you do this, you must move the entire "live/" directOry-not just the library files .)2. Download the newest version of The mplayer source code.3. If, in step 1, you moved the "live/" directory to "/usr/local/", "/usr/LOcal/lib/", or"/usr/lib/", runCD mplayer *;./configure -- enable-liveOthe

specific application to determine the format and meaning of the payload. There is a table in the official documentation that shows the correspondence between the format code and the specific format, and the additional format code may not be defined in the RTP protocol. Sequence number–increments by one for each RTP data packet sent, and May is used by the rece

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

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

real-time audio and video domain UDP is the kingly The Transport layer scheme for real-time audio and video interaction on the Internet has two types: TCP (e.g. RTMP) and UDP (e.g. RTP). The TCP protocol provides 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 correc

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 provide

Directly go to the topic. After JPEG compression, the data is transmitted to the network through the RTP/RTCP protocol. This topic uses the RTP/RTCP protocol stack of jrtplib, first, obtain the source code of the jrtplib package on the internet, decompress the configuration, compile and install the package, and OK if n

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 SIP protocol and some of its applications are worth learning. In this regard, we will explain the configuration of rtp sip today. SIP (Session Initiation Protocol) is usually used for VOIP calls, call establishment, call negotiation, and call termination. it helps two terminals to recognize each other, but it does not process media. When A call is established

3.1Three additional packages: Android. Hardware. USB, Android. MTP, and android.net. RTP! USB, MTP, RTP------- There are three words, all bloody, so people are excited and excited. Why don't you get started with Google? Android. MTP Allow connected camera and other devices to directly use PTP (image transfer protocol) MTP (Media transfer