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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).<br>
So it will be better to keep both streams a little bit under the maximum
allowed to be safe.<br><br>
Cesare<br><br>
At 15.48 01/03/2007, you wrote:<br>
<blockquote type=cite class=cite cite="">The problem with bridging to SIP
is bigger than you think<br>
<br>
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.<br>
Also since I wrote codec_amr and format_amr, i gave chan_zap the amr
capability so the call is transcoded.<br>
Currently when I bridge to SIP it works but there is a BIG problem, it's
the FPS that the softphone is sending.<br>
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.<br>
Also, the softphone is not usually capable of sending a SIP info as Video
Fast Update asking the softphone to reencode an I-frame.<br>
That's why I'm currenty working on the transcoding of H263 <>
H263.<br>
<br>
<br>
-- <br>
Amin Ramtin<br><br>
<br><br>
<br>
<br>
<hr>
<br>
> Subject: Re: [Asterisk-video] H324M AMR audio working!!!<br>
> From: mbrancaleoni@espia.it<br>
> To: asterisk-video@lists.digium.com<br>
> Date: Thu, 1 Mar 2007 14:39:23 +0100<br>
> <br>
> Hi,<br>
> <br>
> On Thu, 2007-03-01 at 13:46 +0100, Sergio Garcia Murillo wrote:<br>
> > From: "matteo brancaleoni"
<mbrancaleoni@espia.it><br>
> > Sent: Thursday, March 01, 2007 1:06 PM<br>
> > > > > > asterisk: symbol lookup error:<br>
> > /usr/lib/asterisk/modules/app_h324m.so:<br>
> > > > > > undefined symbol: TIFFReverseBits<br>
> > I have updated a new verson that should solve the problem
also.<br>
> > Please try it id you can to make sure I haven't broke
anything..<br>
> <br>
> Yes, seems ok :)<br>
> thanks a lot :)<br>
> <br>
> btw... the audio is a bit "broken", I mean that<br>
> has some interruptions in it.<br>
> But maybe is the encoding... I'll check.<br>
> <br>
> When switching between 2 videos, the audio always <br>
> arrives immediately and the video a bit later.<br>
> But maybe this 3G related :)<br>
> <br>
> really good work.<br>
> Now we need only an amr codec and we can bridge sip :)<br>
> <br>
> Matteo<br>
> <br>
> -- <br>
> Matteo Brancaleoni<br>
> R&D Director<br>
> Tel :+39.02.70633354<br>
> Voip :sip:matteo@sip.voismart.it<br>
> <br>
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