rtp

问题 I have an IP camera that is located in a different country ( with a different time zone ) and that has it's own date-time values applied, ( for example:~2012-04-16 11:30:00 ) then the one my PC is located at. ( so my PC's time for example is ~2012-14-16 06:10:00 ) My purpose: When streaming, i need to get this date-time value that is set in camera ("11:30:00") ( I'm not interested in a current local time of my PC ). Is there any way to calculate camera's date-time value from RTP's timestamp ?

问题 I am developing an application based on VOIP(SIP) . I am able to send Invite and Invite-Ack between two emulators but now i would like to send RTPpacket after Invite-ack message in SIP call flow. Is it possible to send like that. If yes, please can anyone help me regarding this. After that i m going to integrate this for both send and receive part of SIP video call. Any help would be appreciated lot!!! Thanks in Advance!!!!! 回答1: i think i will have to look at the RTP implementation of

问题 I am trying to configure the websocket to work with asterisk 11. But there is some issue. The steps I have followed are: In http.conf enabled the following enabled=yes bindaddr=0.0.0.0 bindport=8088 I have also configured the asterisk with DTLS support. But when I try to connect to the websocket new WebSocket ("ws://mySeverIp:8088/ws"); . It throws an error WebSocket connection failed: Error during WebSocket handshake: Unexpected response code: 400 Anyone please help. Thanks 回答1: Here is a

问题 Please help me solve this tricky problem, making me suffering for almost one week. How to make streaming video suffering packet loss? Switch: Pica8 3290 Computer: core i7 2600, 8GB Link: 100Mps Streaming video : RTP (1080P、4K) I've already tried "iperf", "iperf3" and "Packeth" to generate UDP packets. However, these 3 sofwares seem to measure the residual capacity of the link first and then send the amount of the packets fit the capacity. E.g.: (No Video streaming) iperf send almost 100Mps

问题 I'm looking to write a voip application, I know JMF has RTP code but it's old and 32bit only. Is there a robust library that does RTP/RTSP that can be used with java? Thanks 回答1: no way JMF - its 10 years out of date list java RTP RTSP packages: RTSPClientLib.tar.gz on code.google http://github.com/brunodecarvalho/efflux i used the efflux on a java project and it was worth the time learning his packages. The combined netty/efflux is very robust on the variety of RFC's involved in networking

问题 How i can get stream from Ip Camera, Its using RTP, stream is MPEG4, i have multicast address and port,and i have ip camera's IP address and Port Number. And I cant reach via http forexample (http://ip/jpeg) And I cant reach stream with VLC Player too. forexample (rtp://ipadressofcam:port) and ( rtp://multicastaddress:port ) So What is ffmpeg command of that? I have windows OS, I only write code with C# right now. But producer created their own ocx which used for viewing cam,that plugin can

问题 I'm trying to implement RTSP protocol in Java according to http://www.csee.umbc.edu/~pmundur/courses/CMSC691C/lab5-kurose-ross.html example; I have succeed up to successful communication with VLC via RTSP requests and streaming RTP packets; RTP packets with JPEG payload are not recognized by VLC well; that's why I supposed I send malformed RTP. I sniffed them with wireshark and compared them with packets, sniffed from successfull RTSP communication of gstreamer RTSP streamer and VLc. I was

问题 I have to get stream video [ which is from rtsp server] width and height. Third party servers give the following info at RTSP DESCRIBE REQUEST: One RTSP server give me width-height Server Response: RTSP/1.0 200 OK .... Content-Type: application/sdp Content-Length: 376 a=x-dimensions:1280,1024 // GET WIDTH HEIGHT .... a=x-dimensions:1280,1024 But the other does not give me width/height info.... It seems that it supports ONVIF... Server Response: RTSP/1.0 200 OK x-Accept-Dynamic-Rate: 1 ...

问题 I want to read from an RTP stream, but when I specify "test.sdp" to avformat_open_input() I get this message: [rtp @ 03928900] Protocol not on whitelist 'file'! Failed: cannot open input. avformat_open_input() fail: Invalid data found when processing input Normally if I were using ffplay on the console, I would add the option -protocol_whitelist file,udp,rtp and it would work fine. So I tried this: AVDictionary *d = NULL; av_dict_set(&d, "protocol_whitelist", "file, udp, rtp", 0); ret =

问题 From Mozilla site: https://developer.mozilla.org/en-US/docs/Web/API/Media_Streams_API "A MediaStream consists of zero or more MediaStreamTrack objects, representing various audio or video tracks. Each MediaStreamTrack may have one or more channels. The channel represents the smallest unit of a media stream, such as an audio signal associated with a given speaker, like left or right in a stereo audio track." That clarifies what a channel is. Several recent RFCs (E.g. 8108) refer to the need to