<div>Hi Borja , </div>
<div> </div>
<div>Thanks a lot for your response. I looked into i6net products and I will download VXI voiceXML browser today . I also checked another product "Video Coverter" , Do you have any idea , when company is supposed to realase this tool . </div>
<div> </div>
<div>If I get any problem , I will let you know . </div>
<div> </div>
<div>Thanks , </div>
<div>Aman<br></div>
<div class="gmail_quote">On Tue, Mar 30, 2010 at 1:01 AM, Borja SIXTO <span dir="ltr"><<a href="mailto:borja.sixto@i6net.com">borja.sixto@i6net.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hi Aman,<br><br>We had tests to pass video h324m calls with sip audio calls (SIP/RTP alaw) and it works fine too.<br>
You can use a VoIP provider (but without echo cancellation, and without any transcoding) or a front end Asterisk to forward video calls (as a switch/gateway).<br>(We done some modifications in our VXI to convert an Asterisk in a transparent call forwarder for the video : Asterisk normaly convert the audio in PCM...).<br>
<br>My company uses the Sergio's tools through our Asterisk VoiceXML browser and all works fine.<br><br>If you use SIP Video Phone, you will probably need to add the transcoder because most of the soft phones don't be able to respect the H324m video profile (bandwidth limitation and strict h263 video codec).<br>
<br>Regards,<br><br><br>Borja from i6net<br><br><br>Le 30/03/2010 08:49, Klaus Darilion a écrit :
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<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote"><br>Am 30.03.2010 02:12, schrieb aman kumar:<br> <br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hello ,<br>I am a beginner user for asterisk and Linux environment . I need some<br>help for setup video calls and video streaming with asterisk .I will<br>
appreciate , If you can me some links, white paper , examples or<br>any kind of help for that .I have downloaded AsteriskNow and installed<br>in my server. I have following questions , please response.<br>1. Is AsteriskNow good for Video calls.<br>
<br></blockquote>If you only want to make video calls between SIP phones, then it should<br>be OK.<br><br>If you also do video streaming to SIP phones, or if you need video<br>telephony also to 3G UMTS video phones, then you need to patch Asterisk<br>
and I think then it is better to use a plain Asterisk version.<br><br> <br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">2. Do I need any addon / plug in / Hardware for the server for video<br>calls .<br> <br></blockquote>For video calls: no, just enable video in sip.conf (search for video in<br>
the sample config file)<br><br>For 3G UMTS video calls: yes. You need ISDN (or SS7) interface cards<br><br>For streaming (RTSP), recording and playback of video files: yes. You<br>need the applications from<br><a href="https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/" target="_blank">https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/</a><br>
<br> <br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">3. How can we stream some static video contents through Asterisk.<br> <br></blockquote>For example you can store the video as mp4 file. Then use mp4play() to<br>
playback the video to the caller. mp4play() is inlcuded in app_mp4.c<br>(<a href="https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/app_mp4/app_mp4.c" target="_blank">https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/app_mp4/app_mp4.c</a>)<br>
<br> <br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">4. Do we require any type of media server for contents or streaming video.<br> <br></blockquote>You have 2 options:<br>
1. video is provided by Asterisk<br><br>SIP phone<--SIP+RTP--> Asterisk<br> ^ (using mp4play())<br> |<br> video.mp4<br><br><br>2. video is provided by a streaming server (e.g. darwin) and Asterisk<br>
(app_rtsp.c) bridges between the SIP call and the video stream.<br><br>SIP phone<--SIP+RTP--> Asterisk<---RTSP-------> Darwin<br> (using rtsp()) ^<br> |<br>
video.mp4<br><br><br> <br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">5. What are the basic steps for integration Video with asterisk server.<br> <br></blockquote>Hint: only try the enxt step if the previous works<br>
<br>1. Install a "vanilla" (plain) Asterisk<br><br>2. Start with SIP-to-SIP video calls. By 2 webcams and use XLite to make<br>video calls. (enable video in sip.conf)<br><br>3. Try also with the Echo() application. Then the caller should see itself.<br>
<br>4. Add app_mp4.c to Asterisk and reinstall Asterisk<br><br>5. Use mp4save() to record a video call from a SIP phone. Use mpeg4ip<br>tools to analyze the recorded video (audio, video and hint tracks).<br><br>6. Use mp4play() to playback the recorded video to the SIP phone<br>
<br>7. use ffmpeg and mpeg4ip tools to create a video with hint tracks. Then<br>playback this video to the caller using mp4play.<br><br>regards<br>klaus<br><br><br> <br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">I have asked too many questions so that you can come to know , what is<br>my Goal. Please guide me .<br>Thanks ,<br>
Aman<br><br> <br></blockquote> <br></blockquote><br></div></div><font color="#888888">-- <br>Borja Sixto, Research& Innovation - <a href="http://www.i6net.com/" target="_blank">http://www.i6net.com</a><br>Office: +34 911877477 | Gtalk: bsixto | Skype: borja.sixto<br>
<br></font></blockquote></div><br>