rtp

Gstreamer, rtspsrc and payload type

由 折月煮酒 提交于 2019-12-13 15:14:22
问题 I'm having difficulties in retrieving rtsp stream from a specific camera, because the rtp payload type the camera is providing is 35 (unassigned) and payload types accepted by the rtph264depay plugin are in range [96-127]. The result is that gstreamer displays ann error like: <udpsrc0> error: Internal data flow error. <udpsrc0> error: streaming task paused, reason not-linked (-1) Other cameras that I have tested are working because they define a good payload type. FFmpeg, MPlayer and other

搭建直播系统，常见的网络传送协议有哪些？

由 非 Y 不嫁゛ 提交于 2019-12-13 09:10:00
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 直播将我们的沟通方式从传统的二维空间直接进化到了现在的三维模式，最主要的原因就是因为直播具有很强的实时性和互动性功能。在计算机网络中，信息的交换必须按照预先共同约定好的过程进行，才能实现实体间的交互，这就是网络中的传送协议。 如果没有传送协议，传信双方的信息交换就会出现问题。 所以今天拓幻科技就来聊一下在直播系统搭建过程中，可以使用到的一些网络传输协议有哪些？ RTMP 协议: RTMP 是Real Time Messaging Protocol英文的缩写，即实时消息传输协议。RTMP是一种设计用来进行实时数据通信的网络协议，是Adobe Systems公司为Flash播放器和服务器之间音频、视频和数据传输开发的开放协议。由于其一般传输的音视频格式是flv、f4v。web上通过flash播放器播放，pc端大多数浏览器基本支持，但是移动端几乎都不支持，一般主要用来在Flash/AIR平台和支持RTMP协议的流媒体/交互服务器之间进行音视频和数据通信。 它具有多种变种： 1)RTMP 工作在TCP之上，默认使用端口1935； 2)RTMPE 在RTMP的基础上增加了加密功能； 3)RTMPT 封装在HTTP请求之上，可穿透防火墙； 4)RTMPS 类似RTMPT，增加了TLS/SSL的安全功能； 2. RTSP

Receiving UDP in Java without dropping packets

由 ▼魔方 西西 提交于 2019-12-13 07:10:07
问题 I have a library which I need to improve since it is dropping to many packets. I want to receive a RTP-Stream, but the streamer is sending bursts of 30-40 packets within a millisecond (MJPEG-Stream). I can see the packets being complete when monitoring traffic in Wireshark. But when trying to receive them in Java, I lose a lot of those packets. I have already been able to improve the libraries behavior by implementing a ring buffer that would constantly get filled whenever a packet is

Can you recommend a solution to convert real time stream from pc camera to the format of rtp/rtsp?

由 丶灬走出姿态 提交于 2019-12-13 04:22:16
问题 I just walked through this thread, and found html5 video is fantastic. So now what I need to do is convert video streams from pc camera into rtp/rtsp format. Is there a good solution you recommend? 回答1: vlc and ffserver both can do this. 来源： https://stackoverflow.com/questions/3014716/can-you-recommend-a-solution-to-convert-real-time-stream-from-pc-camera-to-the-f

How to decode RTP packets and save it has .wav file

由 ↘锁芯ラ 提交于 2019-12-12 21:25:25
问题 I am trying to develop an application in which a sip call is established and then i am capturing rtp audio packets. As they are encoded so i need to decode them and save it has .wav file. Tried using NAudio but didnt worked. Is there any solution using NAudio or any other source to solve this problem... the code i used is as follow. data is the byte array in which rtp packet data is. System.IO.MemoryStream stream = new System.IO.MemoryStream(data); RawSourceWaveStream rsws = new

wrap h.264 stream in mp.4 container and stream it with nodejs

由 大城市里の小女人 提交于 2019-12-12 03:27:40
问题 I have a stream of h.264 data from a remote webcam. If i save it to a file i'm able to play it in VLC (meaning that the data arrives intact). The final goal is to turn this stream into a virtual webcam. After looking around I found manyCam as a possible solution - therefor i want to serve the h.264 data on a local IP in MP4 format. Two questions: first , I'm trying to wrap the h.264 with the mp4 container using ffmpeg (using fluent-ffmpeg npm library that exposes the ffmpeg API to Nodejs).

Some seconds delay on starting sending Voip with Android.net.rtp

由 女生的网名这么多〃 提交于 2019-12-12 03:24:43
问题 I implemented an Android app that uses Voip by Android.net.rtp library. It simply gets voice from device microphone and sends it in Voip (to another Android or to a PC receiver). The problem is that on some devices the voip trasmission start after 2–3 seconds. I don't mean that there is a delay of 2–3 seconds in delivering packets, I mean that the first 2–3 seconds of voice are not sended. After those 2–3 seconds everything works properly. The strange thing is that it happens only on some

iOS RTP live audio receiving

由 坚强是说给别人听的谎言 提交于 2019-12-12 03:23:25
问题 I'm trying to receive a live RTP audio stream in my iPhone but I don't know how to start. I'm seeking some samples but I can't find them anywhere. I have a Windows desktop app which captures audio from the selected audio interface and streams it as µ-law or a-law. This app works as an audio server that serves any incoming connection with that streaming. I have to say that I've developed an Android app that receives that stream and it works, so I want to replicate this functionality on iOS. In

Changing NALU h.264/avc, for RTP encupsulation

由 醉酒当歌 提交于 2019-12-12 03:22:46
问题 What I can and cant change in NALU in terms of syntex and size, if the nal is meant for RTP encupsulation? 回答1: You can change whatever you want, provided that resulting bit stream is still compliant to: MPEG-4 Part 10 Specification (H.264) RTP RFCs 3550 (RTP), 3984 (RTP Payload for H.264) 来源： https://stackoverflow.com/questions/7560060/changing-nalu-h-264-avc-for-rtp-encupsulation

How does ohrwurm use libpcap and arpspoof to corrupt RTP traffic?

由 跟風遠走 提交于 2019-12-11 19:43:45
问题 I'm trying to evaluate a tool called ohrwurm, which claims to be able to corrupt RTP traffic between two SIP endpoints. By reading its source code I don't believe it works, and would like other's opinions before I try it out. It's premise is simple: Assume endpoint A has IP address 192.168.0.11, and endpoint B has IP address 192.168.0.22. On a third box C on the same subnet as A and B execute the following commands in two SSH sessions: arpspoof 192.168.0.11 arpspoof 192.168.0.22 Execute
订阅 rtp