rtp

Streaming Video From Android

故事扮演 提交于 2019-12-04 07:57:02
问题 I'm trying to stream video from the Android phone, which should be watched in an mediaplayer. I've been looking at http://www.mattakis.com/blog/kisg/20090708/broadcasting-video-with-android-without-writing-to-the-file-system which seems to be a dead end since it send the raw file data, and not a streamable format. Then I tried using some code from SipDroid, more specific; parts of VideoCamera.java, RtpPacket.java and RtpSocket.java, which gives a stream on UDP, however these is not playable

How to stream pcap file to RTP/RTCP stream?

我是研究僧i 提交于 2019-12-04 07:35:20
I have captured three different stream as pcap file with meta datas. How can I stream back to RTP/RTCP stream? If I understand correctly, you have the pcaps, but you want to get the RTP from them? Wireshark UI You could use Wireshark's UI to easily take the RTP from the pcap via the Menu: Telephony/RTP/ then show all streams... click a stream it lists, and then 'analyize.' However, if you want to automate this, and avoid the UI... you can use tshark. I found several tutorials online and used them to build a test harness that automatically rebuilds the audio/rtp on a pcap, then makes a wav and

RTSP Client in Android

青春壹個敷衍的年華 提交于 2019-12-03 22:55:43
问题 The same question might have been asked but I didn't find any solutions available for me. I'm trying to play RTSP stream in android and the server is Darwin streaming server. For now I’ve tried VideoView and MediaPlayer, both of which always work fine when I use 3G but have problems when I use Wifi sometimes, specifically when I use Wifi at my workplace, and the error in LogCat: PlayerDriver( 31): buffering (0) I have googled around and guess the reason maybe the Wifi hotspot is behind the

H264 RTP packet parsing [closed]

两盒软妹~` 提交于 2019-12-03 21:02:58
Closed. This question is off-topic. It is not currently accepting answers. Learn more . Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 3 years ago . I am looking for source code of RTP Payload Format for H.264 Video according to RFC(3984). It would be great if someone can point out it. Also I am looking for wireshark plugin which can show me the packet decoding of h264. Thank you guys in advance. Laurent Etiemble You can start with FFMpeg's source code (under LGPL); the rtp_h264.c contains the code that parse H.264 RTP frame. You should have a

C# - Capture RTP Stream and send to speech recognition

流过昼夜 提交于 2019-12-03 16:32:58
What I am trying to accomplish: Capture RTP Stream in C# Forward that stream to the System.Speech.SpeechRecognitionEngine I am creating a Linux-based robot which will take microphone input, send it Windows machine which will process the audio using Microsoft Speech Recognition and send the response back to the robot. The robot might be hundreds of miles from the server, so I would like to do this over the Internet. What I have done so far: Have the robot generate an RTP stream encoded in MP3 format (other formats available) using FFmpeg (the robot is running on a Raspberry Pi running Arch

Can profile-level-id and sprop-parameter-sets be extracted from an RTP stream?

假如想象 提交于 2019-12-03 16:13:45
I'm trying to stream live video from my android phone to a desktop RTSP server on my PC. The streamed video can be played in another device. I'm using H.264 video encoder, so the SDP returned by the server (as the reply of DESCRIBE request) should contain the profile-level-id and sprop-parameter-sets fields. The Spydroid project shows how to extract these info from a dummy file recorded to SD card by parsing it (from the avcC block). But I cannot do it like that. In Spydroid, the media recorder and the RTSP server are on the same device, so the server can always record a test file with the

Keeping Alive Rtsp Connection

醉酒当歌 提交于 2019-12-03 13:24:18
I have a simple Rtsp Client...The client send Rtsp Commands to Rtsp Server and get RTP packets from server. The problem is that after a time( about 3-4 minute) my 3rd party RTSP Server drop connection with my RTSP Client. I have not implemented RTCP...I take rtp packets from rtsp server but does not send any RTCP PACKET... I make simple search and find that some guys use some RTSP commands[ like OPTIONS, SET PARAMETER-GET PARAMETER ] too keep alive Connections between RTSP Server and Client... But in my case that does not work... Here is my questions: What is the best way to keep alive

How to start RTP stream inside a SIP/SDP call

柔情痞子 提交于 2019-12-03 12:42:46
问题 I've managed to set up a SIP call using the JAIN-SIP API for Java. Now I would like to stream some video once a dialog has been established. I've read that this is possible with SDP and RTP, and I've found multiple examples on how to define a SDP/RTP body in a SIP packet. But once you have negotiated capability etc. on nodes, how do you actually start the RTP stream? Do you start an RTP streaming server outside or inside your Java application? If so, how? What is the link? In what I'm able to

Streaming RTP/RTSP: sync/timestamp problems

人走茶凉 提交于 2019-12-03 11:47:22
问题 I'm having some trouble streaming H.264 video over RTSP. The goal is to live-stream a camera image to an RTSP client (ideally a browser plugin in the end). This has been working pretty well so far, except for one problem: the video will lag on startup, stutter every few seconds, and has a ~4-second delay. This is bad. Our setup is to encode with x264 (w/ zerolatency & ultrafast) and packed into RTSP/RTP with libavformat from ffmpeg 0.6.5. For testing, I'm receiving the stream with a GStreamer

Trouble syncing libavformat/ffmpeg with x264 and RTP

[亡魂溺海] 提交于 2019-12-03 10:48:09
问题 I've been working on some streaming software that takes live feeds from various kinds of cameras and streams over the network using H.264. To accomplish this, I'm using the x264 encoder directly (with the "zerolatency" preset) and feeding NALs as they are available to libavformat to pack into RTP (ultimately RTSP). Ideally, this application should be as real-time as possible. For the most part, this has been working well. Unfortunately, however, there is some sort of synchronization issue: