rtp

问题 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..

问题 I had a RTP Stream socket, receiving a JPEG Stream, from a samsung network camera. I dont know much about how JPEG format works, but i do know that this incoming JFIF or JPEG stream is giving me the JPEG header +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Type-specific | Fragment Offset | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Type | Q | Width | Height | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ and then +-+-+-+-+-+-+

问题 I'm developing a DirectShow filter which has 2 input pins (1 for audio, 1 for video). I'm using libavcodec/libavformat/libavutil of FFMpeg for encoding the video to H264, audio to AAC and mux it/stream using RTP. So far I was able to encode video and audio correctly using libavcodec but now I see that FFMpeg seems to support RTP muxing too. Unfortunatelly, I can't find any example code which shows how to perform H264 encoding and RTP muxing. Does anybody know good samples? 回答1: Try checking

问题 I am looking for the SIP stacks for Android and iPhone. I found plenty of similar questions, which are sometimes quite old... I do not care too much if the solution is commercial (but this is preferred) or open source. So far I found Commercial solution from RADVISION for Android Open source SIPHONE for iPhone LINPHONE which covers both platforms. Gingerbreadhas built-in SIP stack - but seems that it has some limitations and might be removed by MNO... My questions are: Does anybody has good

问题 A video camera is sending me video data as RTPs via UDP on a local port. Does ffmpeg support automatically conversion of input (H.264 payload) to MP4 ? How to do that ? 回答1: This should work: ffmpeg -i udp://localhost:1234 -vcodec copy output.mp4 Or try: ffmpeg -i rtp://localhost:1234 -vcodec copy output.mp4 Replace 1234 with your port. I assume that the input is already in H.264, if not, remove the -vcodec copy. 来源： https://stackoverflow.com/questions/12003014/pipe-udp-input-to-ffmpeg

问题 I want to make a custom sipdroid client by using reverse byte order. I think that makes other Voip clients cannot decode these data. So I read the code of the SipDroid. I found RTP data goes this way: 1. AudioRecord.read(originalPCM) 2. encode(originalPCM, encodedData) 3. rtp_socket.send(rtp_packet) //the encodeData is rtp_packet's data part And the other side is: 1. rtp_receive(rtp_packet) 2. decode(encodeData, PCMData) //the encodeData is rtp_packet's data part 3. AudioTrack.write(PCMData)

问题 From RFC 3550: If a receiver discovers that two other sources are colliding, it MAY keep the packets from one and discard the packets from the other when this can be detected by different source transport addresses or CNAMEs. The two sources are expected to resolve the collision so that the situation doesn't last. In a unicast configuration with one receiver and two senders that only communicate with receiver, how SSRC collisions may be detected by senders? One guess is that receiver should

问题 I've developed an app which sends RTP packets to a local ip client. So the client has to listen on the specified port (rtp://:@portnumber, on VLC) to play the streamed data. Right now i'm going to develop the code needed to create the SDP file needed to start streaming. My doubt is, how to send this file to the client? At the beginning of the RTP stream? Really n00b at this point. Any help will be useful. Thanks 回答1: VLC specifically supports RTSP , HTTP , SAP protocols for establishing

问题 There is a spec that literally requires this: When the SDP negotiation results in the use of the Baseline Profile (BP), a client shall not send Single-Time Aggregation Packet type A (STAP-A) packets, even when the packetization-mode 1 has been negotiated. When accepting the use of the Constrained Baseline Profile (CBP) a client shall support the use of STAP-A packets when packetization-mode 1 was negotiated. can anybody comment that? Doesn't that sound like complete nonsense? How is that even

问题 Hi could someone please explain to me how to achieve a lipsync between two RTP streams (from the same RTSP session). I'm trying to calculate proper pts for ffmpeg AVPacket but I'm missing something and I can't get my head around this. I have the following data available to me: u64RTCP_NTP_TS - NTP timestamp from RTCP Sender Report u32RTCP_TS - Timestamp from the RTCP Sender Report u32AudioRTP_TS - Timestamp from the Audio RTP packet u32VideoRTP_TS - Timestamp from the Video RTP Packet I've