rtp

问题 I have to develop plug-in module which will allow us to stream live video from IP Cams if the IP of the cameras are available. The format and encoding for the same is: MPEG4 RTSP/RTP but I dont know from where to start. I have search in many site, unable to find any thing related. while browsing the Apples doc's, I found AVFoundation is used to stream live video from IP Cam's. UPDATE I found this from GITHUB but dont know How to implement it in my App. Please help me by sample code or related

问题 I try to make a H264 RTP stream from a Raspberry Pi 3 with a camera module to a video tag. Using the following code to start the stream raspivid -t 0 -h 720 -w 1080 -fps 25 -hf -b 2000000 -o - | \ gst-launch-1.0 -v fdsrc \ ! h264parse \ ! rtph264pay \ ! gdppay \ ! udpsink host="192.168.0.11" port=5000 Then I provide a simple webpage with a video tag: <video id="videoTag" src="h264.sdp" autoplay> <p class="warning">Your browser does not support the video tag.</p> </video> The src references

问题 I read the RTP specification and I can't seem to get my head around the RTP Packet Timestamp. I tried to implement it in different ways in my server but I can't get the player to play it right. The "wrong" behaviour I'm having here is that ( I use VLC player to play the RTSP url to my server ) I find the logs of the player saying "buffer way too late" and "wrong PTS value" . Which means somehow that the audio frames arrives in a packets whose timestamp is not set right. I tried to implement

问题 I wrote a C++ application (running on Linux) that serves an RTP stream of about 400 kbps. To most destinations this works fine, but some destinations expericence packet loss. The problematic destinations seem to have a slower connection in common, but it should be plenty fast enough for the stream I'm sending. Since these destinations are able to receive similar RTP streams for other applications without packet loss, my application might be at fault. I already verified a few things: - in a

问题 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

问题 I am new to Android development since a few weeks, and I need to write an app that can display the user a live stream multicasted in UDP or RDP. The stream is located at an address such as "rtp://230.0.0.11:1234", and is emitted by WIFI thanks to this module : http://www.ikusi.es/public/ctrl_public_prod.php?accion=verProducto&id_familia=34&id_gama=186&id_producto=351 I already tried to read it from a player (Daroon player, from PlayStore), and it worked well, so I assume that my foolowing

问题 I have a raw H.264 Stream from an IP Camera packed in RTP frames. I want to get raw H.264 data into a file so I can convert it with ffmpeg . So when I want to write the data into my raw H.264 file I found out it has to look like this: 00 00 01 [SPS] 00 00 01 [PPS] 00 00 01 [NALByte] [PAYLOAD RTP Frame 1] // Payload always without the first 2 Bytes -> NAL [PAYLOAD RTP Frame 2] [... until PAYLOAD Frame with Mark Bit received] // From here its a new Video Frame 00 00 01 [NAL BYTE] [PAYLOAD RTP

问题 I want to transmit(send-only) a .wav file from my android to a softphone (x-lite) so that the called person on x-lite can hear the sound of the .wav file. The scenario is as follows: Android and x-lite are both in the same WLAN and both connected to FreeSwitch. I can call x-lite from the android phone. If the call is accepted on the x-lite the android sends the .wav file and I can see in wireshark that RTP pakets (G.711 PCMU) are send from the phone to x-lite. I can hear some sound but not

问题 I'm writing a rtp video streamer for android that reads h264 coded data from an Android local socket and packetize it. The thing is that I did it but I keep getting black frames in the client side (Voip). The communication goes like this: Android -> Asterisk -> Jitsi (Osx) (and reverse) There are a few things that I haven't understood yet: 1) Android's mediarecorder gives me a raw h264 stream, How can I know when a NAL starts / ends based on that stream? It doesn't have any 0x000001 pattern

问题 I have got one HD video "ed_hd.avi" on System#1. Would like to stream it over network and play the content from System#2. I am using GStreamer on Ubuntu 11.04, tried a lot on this. Variety of errors makes this objective difficult to diagnose. Will be thankful for getting a working command for the System#1-end and System#2-end. What I have tried is as follows: System #1: gst-launch filesrc location=ed_hd.avi ! decodedin ! x263enc ! video/x-h264 ! rtph264pay ! udpsink host=127.0.0.1 port=5000