Hi there, I'm writing custom DirectShow source push filter which is supposed to receive RTP data from video server and push them to the renderer. I wrote a CVideoPushPin class which inherits from CSourceStream and CVideoReceiverThread class which is a wrapper for a thread that receive RTP packets from video server. The receiver thread ...
Hi all, I have started playing around with RTP on Java in Android and was wondering if anyone could give me a bit of help or guidance. Here is the code I have started on: public void rtpTest() throws UnknownHostException, SocketException, RtpException{ RtpManager rtpManager = new RtpManager(myAddress); Log.d("RTPMANAGER", "IPADDRESS ...
If the client wants to watch a stream that is on my RTSP server, it first tries to setup a stream through the UDP protocol. How can I tell it that my server only supports RTP/AVP/TCP and that it should switch transports? I want to terminate the UDP support on my server, but all the clients first try to SETUP the session over UDP, and lat...
Currently, I'm finding a lib able to stream video from multiple sources through one RTP Stream (one connection). Anbody have sugesstion on it? Actually, I figured out that Opal 3.8 is VoIP lib, supported RTP/H264. But I don't know whether it can support mux/demux rtp media from one stream? If no, can you give me some suggesstion? Thank...
Hi Guys, I have a confusion about the timestamp of h264 RTP packet. I know the wall clock rate of video is 90KHz which I defined in the SIP SDP. The frame rate of my encoder is not exactly 30 FPS, it is variable. It varies from 15 FPS to 30 FPS on the fly. So, I cannot use any fixed timestamp. Could any one tell me the timestamp of the...
Hi all! I have a strange problem, mayby you'll help me sort out what's happening. I'm developing a softphone, and sometimes it even works! :) So "sometimes" is the key word. There is a problem with sending voice through microphone over the phone: sometimes it works, sometimes it doesn't. I can't sort out why this is happening. I chec...
I am trying to send rtp audio data from my Android application. I currently can send 1 RTP packet with the code below and I also have another class that extends Thread that listens to and receives RTP packets. My question is how do I continuously send my updated buffer through the packet payload without blocking the receiving thread? ...
We are interested in saving a H.264 stream and replaying it. Is there any one who experience saving h.264 using winpcap and replaying it. We were able to save H.263 and replay, but same logic does not work for H.264. We also tried rtpdump tool to save H264 stream, but we were unable to replay it in that format? thanks in advance ...
Hi, I am developing a walkie-talkie application for Android, and would like to know whether RTP and/or SCTP is natively supported in the latest version of android SDK. From whatever I have dug-up so far, the answer is no. It seems that support for SCTP is coming in JDK 7. But then, I am not sure how this will work with my android de...
Hi, I have a dump of RTP packets of streaming H264 videos that i captured using libpcap. I was wondering if anyone knows of a tool that can generate a playable video file from that. Thanks ...
Hi, I am capturing packets off the network from a video conference HDX. The video is sent in RTP and is encoded in H264. I am trying to capture these packets and generate a video file. I wrote raw H264 data from the packets to disk and i am trying to play it in VLC. VLC just shows a green box. Am i being too naive in my approach with...
Hey folks,i am developing a RTP client on an Android device which can play streaming videos from a server. I am confused regarding how should i start about? i am thinking of developing a web app, using HTML,CSS and Javascript, which can later be wrapped in Android.is this approach correct? does javascript support real time media player?...
Hello, I create a sip session with mjsip to an external voip provider. Then I transmit a test wav file over rtp to the provider using RtpManager. The program runs with no errors and I answer the sip call. However, no audio is transmitted. When I diagnose the network traffic with wireshark, I see a bunch of RTP traffic from my localho...
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 a...
Why do I get a dtmf sound when the E bit is 0 and no sound when it is 1? (RTP packets appear in wireshark either way) Background: I can send out a RFC 2833 dtmf event as outlined at http://www.ietf.org/rfc/rfc2833.txt obtaining the following behaviour when the E bit is NOT set: If for example keys 7874556332111111145855885#3 are press...
Hi there, am relatively new to JMF but have gone through the documents and have a sufficient understanding of how it works. That been said am having some trouble implementing a the server side for RTPSockets. After looking at their illustrations and example. I am still abit confused. Am I to develop a datasource and also datasink class...
Hello to everyone, I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is ...
When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong. I took pipelines from gstrtpbin documentation page and slightly mo...
Hello 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 mu...
I am using SharpPCap which is built on WinPCap to capture UDP traffic. My end goal is to capture the audio data from H.323 and save those phone conversations as WAV files. But first thing is first - I need to figure out what my UDP packets are crossing the NIC. SharpPCap provides a UdpPacket class that gives me access to the PayloadDat...