rtsp 554

Live555 receive RTSP live stream, convert HTTP Live streaming (iOS live) protocolThe RTSP protocol is also a widely used live/on-demand streaming protocol that previously implemented a program that received the RTSP protocol via live555 and then converted it to the HLS (Http live streaming) live protocol file. In order to receive the remote device or the server's

= strrecvbuf.substr (Nindexofnoncevaluestart + Nnoncelen, Nindexofnoncevalueend- Nindexofnoncevaluestart-nnoncelen);char* pszalg = "";char* pszqop = "";char* psznoncecount = NULL;char* pszcnonce = NULL;Char sznonce[128] = {0};sprintf (sznonce, "%s", Strnoncevalue.c_str ());Char szusername[128] = {0};sprintf (szUserName, "%s", "admin");Char szrealm[32] = {0};sprintf (Szrealm, "%s", Strrealmvalue.c_str ());Char szpassword[32] = {0};sprintf (Szpassword, "%s", "admin12345");Char pszmethod[32] = {0}

int main (int argc, char **argv) { Iplimage *pframe = NULL, *srcimage=null; Cvcapture *pcapture = NULL; Pcapture = Cvcapturefromfile ("Rtsp://admin:12345@192.168.7.45:554/h264/ch1/main/av_stream");Pcapture = Cvcreatefilecapture ("rtsp://admin:default@10.1.10.173/h264");Pcapture = cvcreatecameracapture (1);if (!pcapture) {printf ("Can not get the" ' video stream

[GO] Streaming Media protocol introduction (RTP/RTCP/RTSP/RTMP/MMS/HLS)http://blog.csdn.net/tttyd/article/details/12032357RTP reference Documentationrfc3550/rfc3551Real-time Transport Protocol) is a Transport layer protocol for multimedia traffic on the Internet. The RTP protocol details the standard packet format for transmitting audio and video over the Internet. RTP protocols are commonly used in streaming media systems (with the RTCP protocol), vi

transmit multimedia data over an IP network. RTSP provides a scalable framework that enables real-time data, such as audio and video, to be controlled and on-demand. Data sources include field data and data that is stored in a clip. The protocol is designed to control multiple data-sending connections, providing a way to select Send channels, such as UDP, multicast UDP, and TCP, and provide a way to select the RTP-based send mechanism.RTSP (Real time

protocol called Secure Real-time Transport Control Protocol (secure RTCP or SRTCP); The secure real-time transport Control Protocol provides similar security-related features for real-time transport control protocols, just as the secure real-time transport protocol provides for real-time transport protocols.It is optional to use the real-time transport protocol or the real-time transport control Protocol to make it possible to not use secure real-time transport protocols or secure real-time tra

serial number. Timestamp: 32 bits, timestamp. Indicates the sampling time of the first byte in the information packet carried by RTP packet. SSRC: 32 bits, data source ID. Each data stream in an RTP session should have a different SSRC. CSRC list: 0 to 15 items. Each source ID is 32 bits, which contributes to the data source ID. It is valid only when mixer exists. For example, if a multi-channel speech stream is merged into a single-channel speech stream, the SSRC of each original audio channel

The similarities and differences between RTSP, RTMP, and HTTPCommon:1:RTSP RTMP HTTP is applied at the application layer.2: Theoretically RTSP rtmphttp can do live and on-demand, but generally do live with RTSP RTMP, do on-demand http. Video conferencing when the original SIP protocol, and now basically replaced by the

Reprint: http://easydarwin.org/article/Streaming/141.htmlThe similarities and differences between RTSP, RTMP, and HTTPCommon:1:RTSP RTMP HTTP is applied at the application layer.2: Theoretically RTSP rtmphttp can do live and on-demand, but generally do live with RTSP RTMP, do on-demand http. Video conferencing when the

incomingconnectionhandlerrtsp to the task scheduling Env. taskscheduler (). turnonbackgroundreadhandling (frtspserversocket, (taskscheduler: backgroundhandlerproc *) incomingconnectionhandlerrtsp, this );} void RTSPServer::incomingConnectionHandlerRTSP(void* instance, int /*mask*/) { RTSPServer* server = (RTSPServer*)instance; server->incomingConnectionHandlerRTSP1();} void RTSPServer::incomingConnectionHandlerRTSP1() { incomingConnectionHandler(fRTSPServerSocket);} When receiving a connec

Recently in the development of dm368, intends to transplant a gst-rtsp-server on 368. First on the computer to toss a day, finally can run.Gstreamer-0.10 was previously installed on my virtual machine (the GStreamer version is too new and some plugins are not available). GStreamer and Base,good,ugly,bad related plug-in installation did not encounter any problems. Here are the problems I encountered in compiling gst-

Transferred from: http://www.zhihu.com/question/20621558 Yang HuaLinks: http://www.zhihu.com/question/20621558/answer/15661190Source: KnowCopyright belongs to the author. Commercial reprint please contact the author for authorization, non-commercial reprint please specify the source.As a field of expertise, I decided to update this answer.Copyright reserved, not commercial, reprint must be at the beginning of the location of the author, source.With the impression of completion, errors and omissi

= "$LD _library_path:/usr/local/ffmpeg/lib/" souce/etc/profile 4. Download opencv2.4.13, after decompression into the directory, the author is using opencv-2.4.13, execute the following command CD ~/opencv-2.4.13/ mkdir build CD The following make command because the author installs the Cuda, therefore inside uses cuda_generation=auto this compilation option, does not use Cuda compiles the child shoe to be possible to delete this piece configuration to compile. cmake-d cmake_build_type=release

RTP/RTCP/RTSP/SIP/SDP RelationshipRTP (real-time transport protocol, Transport layer)Real-time Transport Protocol) is a Transport layer protocol for multimedia traffic on the Internet. The RTP protocol details the standard packet format for transmitting audio and video over the Internet. RTP protocols are commonly used in streaming media systems (with the RTCP protocol), video conferencing and a Push-to-talk system (with either a/p or SIP), making it

1. s60 uses a multimedia framework (MMF) for video and audio playback and recording. It has a plug-in architecture and can use multiple types of plug-ins for media playback and recording, for example, the RealPlayer engine is a plug-in for the MMF controller and supports video and audio playback and stream. 2. The s60 built-in player uses the RealPlayer engine. HTTP streaming media is not supported, because all video data must be read into the cache at one time before display. 3. the common form

With the arrival of 3G, the bandwidth is higher and the traffic fee is lower, and multimedia applications such as mobile phone and TV will surely develop greatly. I will sum up my previous experience, I will discuss with you how to create a VoD solution for mobile phones, and finally provide a preliminary client implementation result. Welcome to the discussion. First, let's talk about the architecture. For ease of management and expansion, bandwidth restrictions, and multi-user concurrency, comm

Recently, due to the need to implement RTSP-based transmission RM rmvb, no existing solutions have been found on the Internet. Only one live555 has implemented the RTSP service, but only supports RTP transmission of mpg, MP3, WAV and other formats. rm seems very unpopular in the Open Source Field, probably because the formats and protocols are private and not open. Then we started to develop rmvb over

, quality reduction; second, advanced and efficient compression algorithms.Basic Principles The process of stream transmission is generally as follows: (1) After selecting a first-class media service, the web browser and the Web server use HTTP/tcp to exchange control information, in order to retrieve the real-time audio/video stream data to be transmitted from the original information. (2) The web browser on the client starts the Client Program (some playing programs) and uses http to retrieve

Atitit onvif protocol Get RTSP address play Java language ATTILX summary 1.1. Get the RTSP address algorithm and Flow 1 1.2.Onvif camera discovery, WS discovery Mechanism, using XCF class library 1 2. Call Getstreamuri get RTSP address, use class library onvif Java Library by Milgo2 2.1. The problem why use this get address can not play 4 1.1. algorithms and

HTTP (Hypertext Transfer Protocol), RTSP (Real time Streaming protocol live streaming protocol), RTMP (Routing Table Maintenance Protocol Routing Tables Maintenance Protocol) is the application layer protocol, Theoretically all can do live, on-demand, in fact, live more than rtmp and RTSP, on-demand is more use RTSP and HTTP.First, common areas: HTTP (HTT