rtp | 易学教程

H264的RTP载荷格式

阅读更多关于 H264的RTP载荷格式

H264中对于数据的存储分为两层，分别是 VCL: video coding layer 视频编码层这是h264中block, macro block 以及 slice级别的定义，目的是为了独立网络传输，进行高效的编解码 NAL: network abstraction layer 网络提取层 NAL层的不同是为了根据不同的网络把VCL数据进行打包成不同的格式，进行传输 NAL H264中的数据分组有头部(譬如: pkt->data)，解码器可以很方便的检测到NAL的分界，依次去除NAL进行解码，但为了节省码流，NAL没有在数据头部设立特定元素，如果编码数据存储在介质上，NAL依次紧密相连(这时就无法通过对象取读取一个nal数据了)，就无法区分出不同的nal单元，为了解决该问题，存储的时候会在每一个nal前添加0x000001或0x000000。但数据内部也可能出现0x000001或0x000000，为此nal中规定不能出现0x000001或0x000000，源数据需要进行如下变换: 原数据替换数据 0X000000 0X00000300 0X000001 0X00000301 0X000002 0X00000302 0X000003 0X00000303 在网络传输过程中， NAL中的数据被分成NALU(nal unit)单元，然后由RTP进行封装传输如上图所示

Stream multicast media use Gstreamer and VLC

阅读更多关于 Stream multicast media use Gstreamer and VLC

问题 I am writing an application streaming media file ( H264, fortmat I use now is *.MP4 ). I use Gstreamer with pipeline : gst-launch-1.0 filesrc location=/big_buck_bunny_720p_h264.mp4 ! qtdemux name=d d. ! h264parse ! rtph264pay config-interval=3 pt=96 mtu=1024 ! udpsink host=224.1.1.1 port=5000 And I use the following SDP file to receive stream: c=IN IP4 224.1.1.1 m=video 5000 RTP/AVP 96 a=rtpmap:96 H264/90000 In PC I can receive the stream and it playback normally. But now I want to receive

Why RTP/RTSP meddle with my H.264 NALs?

阅读更多关于 Why RTP/RTSP meddle with my H.264 NALs?

问题 I looked in The RFC and noting could explain why the following happens(Though the decoder can still produce the original movie). I transmitted the H.264/AVC nals using VSS h.264 encoder, the byte stream looked something like this E5 46 0E 4F FF A0 23... when I read the movie data one the receiver side after the RTP Broadcaster/RTSP receiver, I get extra unknown data but always in the same places, 8 bytes are added before Start Code prefix (0x00000001), and 2 bytes are added after Start Code

Detecting codec used in RTP stream (for dynamic PTs)

阅读更多关于 Detecting codec used in RTP stream (for dynamic PTs)

问题 Is it possible to detect the codec used in an RTP stream by analyzing the RTP stream alone? I know about the payload type (PT) field in the RTP header -- that can be used to identify codecs that have statically assigned PT numbers. What about the other codecs that use the dynamic PT number range? Are there any heuristics to detect the type of the payload? I'm interested in all kinds of solutions, even if the detection works for one codec only. 回答1: Sure, you can sniff around the packet to see

Broadcast audiostream to multiple devices

阅读更多关于 Broadcast audiostream to multiple devices

问题 I am using android AudioStream to communicate between 2 android devices on wifi, both ways. Is there any way to broadcast an audio message on multiple devices, in the same time ? Is using multiple AudioStreams, one for each device, a possible way? I need to broadcast to 40 receivers. Will the message be delayed if I use multiple AudioStreams ? Do you know other solutions ? 回答1: I think the simplest way is to stream all devices using AudioGroup, you just need to create separate AudioStream for

How to Convert Rtp Multicast Stream from DVB-T to HLS?

阅读更多关于 How to Convert Rtp Multicast Stream from DVB-T to HLS?

问题 I use Mumudvb to get signal from dvb-t and dvb-s to RTP Multicast stream and successfully do that, The result stream URL is something like rtp://239.1.2.1:60001. Now i want to know How can i convert RTP (Or UDP)stream to Http Live Stream (HLS)? Edit : I could convert live stream with ffmpeg, but its not stable, when an error occurred in ffmpeg the conversation stop and there is no way to found fail and for example restart ffmpeg, I am looking for new way for that convert operation. Thanks a

Open Source .net C# library for Real Time transport Protocol

阅读更多关于 Open Source .net C# library for Real Time transport Protocol

问题 Is there good open source library for RTP in .net? My intention is to use in Audio and Video Synchronization problem and to increase Frames per second rate. 回答1: I don't know much about RTP, but you may want to have a look at this article. Also, a Google search for "RTP C#" may help you a bit. Another result from that search includes this Stack Overflow answer. 来源： https://stackoverflow.com/questions/8358230/open-source-net-c-sharp-library-for-real-time-transport-protocol

Playing RTP using VLC

阅读更多关于 Playing RTP using VLC

问题 I currently have an Android application that is capturing video from the camera and streaming it over RTP. I do not use RTSP. I have created a SDP file that describes the RTP feed. I can open the SDP file in Quicktime 7 and can see the stream, attempting to open the SDP file in VLC does not work. I get the following error: live555 demux error: no data received in 10s, aborting I am monitoring the RTP packets using Wireshark and can see the packets on the network (see example below): 281956

Combining an audio and video stream using gstreamer [closed]

阅读更多关于 Combining an audio and video stream using gstreamer [closed]

问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 6 years ago . I am streaming an mp4(mpeg-4) file from one device to another using gstreamer over RTP stream. Basically I am splitting up the mp4 file into its audio and video file and then sending it all to the other device where it gets streamed. Now, I want to save the mp4 file to disk in the other device, but my problem is

How to stream pcap file to RTP/RTCP stream?

阅读更多关于 How to stream pcap file to RTP/RTCP stream?

问题 I have captured three different stream as pcap file with meta datas. How can I stream back to RTP/RTCP stream? 回答1: 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