rtp tutorial

ubuntu9.10 Install OpenSER and use RTP Proxy to realize turn, solve the problem that symmetric NAT brings to SIP voice communication Reprinted from Link http://hi.baidu.com/zj8la8la/blog/item/d700d8b2c11a41abd9335af9.html Leave a note, convenient other people, online resources are not complete, I do a complete bar, at least I test through:My goal is very simple, just realize the SIP network MySQL database support.Start: Set the domain name * If your

1 EVRC Protocol Evolution: 3558->4788->5188 EVRC0 net charge format see 3558 EVRC-WB net charge format see 5188 EVRC codec algorithm based on loose code excitation linear prediction (RCELP) algorithm, with linear pre-and error control functions, EVRC family of various codes see the following diagram: 2 Multi-frame format 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |

sent to the UDP: // 233.233.233.223: 6666 address. ffmpeg -re -i chunwan.h264 -vcodec mpeg2video -f mpeg2video udp://233.233.233.223:6666 1.4. Play the MPEG2 bare stream Specify-vcodec as mpeg2video. ffplay -vcodec mpeg2video udp://233.233.233.223:6666 2. rtp2.1. send the H.264 bare stream to the multicast address. The following command sends the H.264 raw stream "Chunwan. h264" to the address RTP: // 233.233.233.223: 264 ffmpeg -re -i chunwan.h26

Today, I found a strange problem. I can call the SIP client of the lower computer by using the Linphone client of the upper computer to work normally, but in turn there is a problem. Packet Capture found that Linphone sent a large number of IP fragmentation data packets, so google knows that when the data found is larger than MTU, it will generate IP fragmentation data packets. I have already split the RTP package? This should not happen normally. Lin

RTP encapsulation of H.264 (lower) 3. RTP encapsulation implementation 3.1 encapsulation Flowchart 4. RTP solution encapsulation implementation 4.1 Flowchart 5. Summary Hope you are correct! 6. References [1] schuzrinne H, casner S, Frederick R, et al. rfc3550 RTP: A transport protocol for r

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 protocol ). Keep the device connected. The upper-layer app can rec

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 protocol ). Keep the device connected. The upper-layer app can rec

There are three different basic loads (single Nal, non-interleaved, interleaved) in RTP of h264) The application can use the first byte for recognition. The properties of this session are also described in SDP. SDP parameters The following describes how to represent an H.264 stream in SDP: . "M =" the media name in the row must be "video" . The encoding name in the "A = rtpmap" line must be "h264 ". . The clock frequency in the "A = rtpmap" row must

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

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

RTCP RTP protocol format Analysis 1: The task of sequential RTCP is to provide feedback on the delivery quality of the data streams transmitted over RTP, such as packet loss rate, jitter, bandwidth, and rate; when it is detected that the quality is poor, it will make some adjustments based on its own settings when the next packet is sent to achieve optimization. This process is not reflected by the recipie

following command reads the data from the local camera, encoded as MPEG2, and sent to the UDP: // 233.233.233.223: 6666 address. [Plain]View plaincopy FFmpeg-Re-I Chunwan. h264-vcodec mpeg2video-F mpeg2video UDP: // 233.233.233.223: 6666 1.4. Play the MPEG2 bare stream Specify-vcodec as mpeg2video. [Plain]View plaincopy Ffplay-vcodec mpeg2video UDP: // 233.233.233.223: 6666 2. rtp2.1. send the H.264 bare stream to the multicast address. The following command sends the H.264 raw stream "Chun

I have written an article "converting FLV stream to standard h264 and ACC in rtmp", link address Http://www.cnblogs.com/chef/archive/2012/07/18/2597279.html . The extraction of h264 from rtmp is analyzed. In flash projects with audio/video interaction, the audio encoding can only be in speex format. This articleArticleIt is divided into three parts. These are the audio interfaces provided in flex, The speex data in rtmp, and how to convert them to RT