[Asterisk-video] MP4Play async Audio / Video
Sergio Garcia
sergio.garcia at fontventa.com
Tue Sep 25 04:25:55 CDT 2007
You coulde try with qt encoder or the helix mobile producer..
---------- Original Message ----------------------------------
From: "Thomas Z." <mobilemail at gmx-topmail.de>
Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
Date: Tue, 25 Sep 2007 10:52:07 +0200
>Hello Sergio,
>
>do you know a freeware or cheap CBR encoder?
>
>Regards
>Thomas
>
>Sergio Garcia Murillo schrieb:
>> Hi Thomas, Thomas and Ratmin.. :)
>>
>> Back to work form a long abroad weekend (nice Belgium!)
>>
>> As Ratmin has stated the Skew indication would help to indicate the
>> handset the delay between audio and video.
>> It's usually used to correct fixed delays introduced by internal
>> jitter buffers and thinks alike. The problem that Thomas
>> Z. is suffering seems to be produced by using ffmpeg for video
>> encoding (as Thomas without Z points :)
>> The problem is the following, ffmpeg is a great VBR (variable bit
>> rate) encoder, but I've not been able to behave like a
>> CBR (constant bit rate) encoder. The problem is that it's prepared for
>> movies, not for streaming, and defenitevilly not
>> at this low bitrates.
>> The inner loop of the encoder always tries to encode ALL macroblocks
>> in a frame, so in I frames and at a very low
>> bitrate it simply just can't get a small amount of data for that frame
>> and when it's enqued into h324m it takes more
>> than one frame time to send (sometimes even more than a second), the
>> following frames are small and sent after
>> that until the queue is empty, which causes a fast-forward effect
>> on some handsets. As you can imagine adjustime
>> audio to that scenary is jus impossible (even with the Skew Indication
>> the result would be horrible!!)
>>
>> The good solution: use a CBR encoder :)
>> The not so good solution: drop packets when the queue is filled, but
>> you'll end up dropping all the frames between
>> each I frame.. but in that case it would be much better to lower the
>> video fps...
>>
>> BR
>> Sergio
>>
>> ----- Original Message -----
>> *From:* Ramtin Amin <mailto:keytwho at hotmail.com>
>> *To:* Development discussion of video media support in Asterisk
>> <mailto:asterisk-video at lists.digium.com>
>> *Sent:* Monday, September 24, 2007 2:01 PM
>> *Subject:* Re: [Asterisk-video] MP4Play async Audio / Video
>>
>>
>>
>>
>> RTSP has timestamp... so you can usually know how to sync the
>> audio/video
>> But the problem would be more that we'll need to add a Skew
>> Indication concept inside asterisk's core so it could make the
>> message go throught the different channel/application, as it is
>> currently done for VideoFastUpdatePicutre... Which means that
>> currently, if a SIP channel receives a VideoFastUpdatePicture, it
>> is capable of resending it to an other channel...
>>
>>
>> ------------------------------------------------------------------------
>>
>> > Date: Mon, 24 Sep 2007 13:48:54 +0200
>> > From: mobilemail at gmx-topmail.de
>> > To: asterisk-video at lists.digium.com
>> > Subject: Re: [Asterisk-video] MP4Play async Audio / Video
>> >
>> > Hello Ramin,
>> > hello Thomas.
>> >
>> > I think both ways are possible. But I don't know if it easy to
>> implement.
>> > Sergio, what do you think about it?
>> >
>> > Is this also a problem if we stream the audi/video via rtsp?
>> >
>> > Regrads
>> > Thomas
>> >
>> >
>> > Ramtin Amin schrieb:
>> > > hello
>> > >
>> > > The way I solved this problem was by sending a H223SkewIndication
>> > > message to the other terminal
>> > > Acutally, You will have to ask Sergio to add a H245 indication
>> message
>> > > of type h223SkewIndication and when playing the video with
>> mp4play, he
>> > > will have to see the difference of timing between Video and
>> Audio and
>> > > then send this value to the remote terminal so the lip sync
>> would work...
>> > >
>> > >
>> > >
>> > > **
>> > >
>> > > *2.3 Multipoint Lip Synchronization*
>> > >
>> > > In a multipoint VC, each terminal may transmit different
>> > >
>> > > //
>> > >
>> > > /H223SkewIndication /message for associated video and
>> > >
>> > > audio channels in H.223 protocol. To enable lip
>> > >
>> > > synchronization at receiving terminals, MCUs will
>> > >
>> > > transmit accurate /H223SkewIndication /messages. MCUs
>> > >
>> > > may accomplish this by adding delay to equalize the
>> > >
>> > > audio/video skew for all transmitting terminals. When
>> > >
>> > > switching between broadcasting terminals, H.223 may
>> > >
>> > > transmit a new /H223SkewIndication /message reflecting the
>> > >
>> > > audio/video skew of the current broadcaster.
>> > >
>> > >
>> > > --
>> > > Ramtin Amin
>> > >
>> > >
>> > >
>> > >
>> > >
>> ------------------------------------------------------------------------
>> > >
>> > > > From: thomas.frieling at viif.de
>> > > > To: asterisk-video at lists.digium.com
>> > > > Date: Mon, 24 Sep 2007 11:48:12 +0200
>> > > > Subject: Re: [Asterisk-video] MP4Play async Audio / Video
>> > > >
>> > > > Hi Thomas Z!
>> > > >
>> > > > I think this problem is due to the bitrate restriction on 3G
>> calls. The
>> > > > audio stream is always sent immediately while the video
>> stream has to
>> > > > use what is left of the bandwidth. This is why videos become
>> synchrous
>> > > > again when the video bitrate is pretty low for a while in
>> the video...
>> > > >
>> > > > Take a look at this discussion:
>> > > >
>> > >
>> http://lists.digium.com/pipermail/asterisk-video/2007-September/001257.html
>> > > >
>> > > > One solution is to reencode every video with a low bitrate
>> and use a
>> > > > fixed bitrate encoder (ffmpeg is dynamic bitrate for
>> example). I had the
>> > > > impression though that this still doesn't always work,
>> especially when
>> > > > the UMTS connection is not too good...
>> > > >
>> > > > My idea is that we check each time before sending a keyframe
>> if there is
>> > > > already a new keyframe in the queue. If this is the case, we
>> jump to the
>> > > > most recent keyframe and just drop the data before that.
>> > > >
>> > > > What do you think about this? How hard to implement?
>> > > >
>> > > > Regards,
>> > > > Thomas F
>> > > >
>> > > >
>> > > >
>> > > > Am Montag, den 24.09.2007, 11:15 +0200 schrieb Thomas Z.:
>> > > > > Hello,
>> > > > >
>> > > > > We have the problem, that a converted mp4 file is
>> asynchronous via 3g
>> > > > > network.
>> > > > > If we play the file on a pc, everything is synchron.
>> > > > > We tried already to reduce the quality and framerate of
>> the video. But
>> > > > > nothing helps.
>> > > > >
>> > > > > Is it a problem with mp4play or with the video?
>> > > > >
>> > > > > What can we do to get the audio and video synchronized via
>> 3g network?
>> > > > >
>> > > > > Thank you,
>> > > > > best regards
>> > > > > Thomas
>> > > > >
>> > > > >
>> > > > >
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>> > > > --
>> > > > www.ViiF.de - your Mobile Video Community
>> > > >
>> > > > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>> > > >
>> > > > Thomas Frieling - IT Development
>> > > > ViiF Mobile Video GmbH, Poststr. 21-22, 10178 Berlin
>> > > > Cell: +49 (0) 173 63 62 62 3
>> > > >
>> > > > mailto:thomas at ViiF.de
>> > > >
>> > > > Sitz: Berlin, Amtgericht Berlin-Charlottenburg, HRB: 108350B
>> > > >
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>> > > >
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