JMF stream over RTP and VLC

Hi All!
I wrote an application that streams video (without audio) using h263 codec over RTP.
Does anybody know how to play this stream in VLC?
Second issue I am struggling with is to make SDP file that will describe this stream. Any help?
Thank you.
Edited by: 814312 on 22.11.2010 2:05

814312 wrote:
Does anybody know how to play this stream in VLC?Sounds like a VLC question, not a JMF one...
Second issue I am struggling with is to make SDP file that will describe this stream. Any help? Never heard of a SDP file, so I'd assume this isn't really a JMF question either ;-)

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    I've got a question about RTP. I'm trying to make a screen cast applycation using the Robot java class. So I take screenshots to make a new video (avi) file with them using JMF. But I want to send that video using RTP in real time. In fact I want to make screencast over RTP, in real time. Is it possible?
    At this time, I'm able to record the screen, then I save that in a .avi File. When the .AVI file is made, I send it over RTP, but this is not real time.
    Thx
    Flo

    Presumably, you're using the following sample code to create the AVI:
    [http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/JpegImagesToMovie.html]
    And you're using the following sample code(or something similar) to send the AVI via RTP:
    [http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/VideoTransmit.html]
    Regardless, instead of creating a DataSource from a MediaLocator that points to your AVI file, you can just use the DataSource that's generating the AVI files, and make that be the DataSource that's driving your RTP session.

  • Video streaming over UDP

    Hi,
    Does anyone know how to stream a video file, that includes an audio and video codec(from any kind), to a remote client using UDP/RTP ?
    I don't really know how to synch the audio/video over RTP.
    I know this is a hard one, but I'm searching for a guru that knows how do it.
    Thanks,
    em.

    First off, I don't recall if Java has an API or any access at all to QoS mapping.
    UDP is not reliable, so unless you have some sort of packet integrity protocol on your app, then TCP is a much better way to go.
    When you say Codec, are you transporting the codec along with the stream or is that something your client has - or download seperately - as a codec (and you don't need the encoder part to playback on the client) is something extra to the app.
    If you are multicasting a single file such as AVI, WMV, etc. then just transport the file.
    If you are moving seperate streams for the audio, video, and data. Then create three sockets, and create a sync header in the data payload of each, and then the client will recombine the streams' packets to match each sync header.

  • Does Cisco Media Services Interface mark RTP and Signaling packets regardless of Cisco Prime Collaboration/MediaNet Deployment?

    Hi there,
    I'm deploying Cisco UC Integration Lync 9.2(5). There's a QoS portion which states that you can install Cisco Media Services Interface on the user's PC and is to be used with Cisco Prime Collaboration or MediaNet:
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucilync/9/CUCI_BK_C0545A41_00_cisco-uc-integration-for-microsoft/CUCI_BK_C0545A41_00_cisco-uc-integration-for-microsoft_chapter_010.html#CUCI_CN_IDBE72E9_00
    If we don't have Cisco Prime Collaboration or MediaNet, does Cisco Media Services Interface mark RTP and Siganling packets on the outbound regardless? Or do I have to manually set the DSCP values for the PC?
    Thanks,
    John

    When you were placing the internal test calls, where was your phone located?   The RTP streams are usually only set up between the two endpoints (assuming no transcoder or MTP is used).
    Have you have already checked calls between IP phones at your central site (where Unity is located) and IP Phones at site A?   Are these working fine?   If not, then it's likely you do have a QoS issue over the WAN to Site A.
    Have you confirmed whether or not calls from the PSTN gateway at site A experience any voice QoS issues when transferred to an IP Phone at the central site?
    Is the same codec is being used for the MGCP gateway as for IP Phones at that site (check the Region settings for the Device Pool it's in)?
    If these things have all been checked, then you may want to consider collecting a simultaneous sniffer trace from each end of a single call, and comparing what's seen at the two endpoints.
    Kind regards,
    Craig

  • How do I create a JMF streaming server with QuickTime Web Client?

    Can anyone give me some pointers on how to create a JMF streaming server and user can type in a rtsp URL on the web browser and be able to view the streaming video in QuickTime player?
    thanks

    JMF does not have RTSP server capabilities built-in. You would have to program the RTSP request-response protocol yourself.

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