[Asterisk-video] H324M AMR audio working!!!

[Ramtin Amin]

Basically if the softphone is sending H263 way Higher than 50kb, i have to transcode ! and that was my point.
So i was currently doing this job with ffmpeg...
 


Date: Thu, 1 Mar 2007 16:02:25 +0100To: asterisk-video at lists.digium.comFrom: cadm at tiscali.itSubject: RE: [Asterisk-video] H324M AMR audio working!!!It's normal, Amin: for 3G calls you can't exceed 51200 bit/s for the video stream and 12200 bit/s for the audio stream, as stated in the H324M draft(s).So it will be better to keep both streams a little bit under the maximum allowed to be safe.CesareAt 15.48 01/03/2007, you wrote:
The problem with bridging to SIP is bigger than you think My H324m stack is directly implemented into chan_zap, so I can place 3G call with dial(zap/gv1/number) or I receive video call from a cellphone in H263 mode.Also since I wrote codec_amr and format_amr, i gave chan_zap the amr capability so the call is transcoded.Currently when I bridge to SIP it works but there is a BIG problem, it's the FPS that the softphone is sending.Basically, if the H263 video coded by the softphone is about 50kb/s its fine. but if it's bigger (which is usually the case) then you loose frame.Also, the softphone is not usually capable of sending a SIP info as Video Fast Update asking the softphone to reencode an I-frame.That's why I'm currenty working on the transcoding of H263 <> H263.  -- Amin Ramtin 

> Subject: Re: [Asterisk-video] H324M AMR audio working!!!> From: mbrancaleoni at espia.it> To: asterisk-video at lists.digium.com> Date: Thu, 1 Mar 2007 14:39:23 +0100> > Hi,> > On Thu, 2007-03-01 at 13:46 +0100, Sergio Garcia Murillo wrote:> > From: "matteo brancaleoni" <mbrancaleoni at espia.it>> > Sent: Thursday, March 01, 2007 1:06 PM> > > > > > asterisk: symbol lookup error:> > /usr/lib/asterisk/modules/app_h324m.so:> > > > > > undefined symbol: TIFFReverseBits> > I have updated a new verson that should solve the problem also.> > Please try it id you can to make sure I haven't broke anything..> > Yes, seems ok :)> thanks a lot :)> > btw... the audio is a bit "broken", I mean that> has some interruptions in it.> But maybe is the encoding... I'll check.> > When switching between 2 videos, the audio always > arrives immediately and the video a bit later.> But maybe this 3G related :)> > really good work.> Now we need only an amr codec and we can bridge sip :)> > Matteo> > -- > Matteo Brancaleoni> R&D Director> Tel :+39.02.70633354> Voip :sip:matteo at sip.voismart.it> > _______________________________________________> --Bandwidth and Colocation provided by Easynews.com --> > asterisk-video mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-video

Avec Windows Live Spaces, publiez directement des messages électroniques sur votre blog ou ajoutez-y des photos, des blagues et d'autres infos. C'est gratuit ! _______________________________________________--Bandwidth and Colocation provided by Easynews.com --asterisk-video mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-video
_________________________________________________________________
Essayez Live.com, votre nouvelle page d'accueil ! Personnalisez-la en quelques clics pour retrouver tout ce qui vous intéresse au même endroit.
http://www.live.com/getstarted
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20070301/a17adbf9/attachment-0001.htm