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<body>Basically if the softphone is sending H263 way Higher than 50kb, i have to transcode ! and that was my point.<BR>
So i was currently doing this job with ffmpeg...<BR>
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Date: Thu, 1 Mar 2007 16:02:25 +0100<BR>To: asterisk-video@lists.digium.com<BR>From: cadm@tiscali.it<BR>Subject: RE: [Asterisk-video] H324M AMR audio working!!!<BR><BR>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 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>
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<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>> _______________________________________________<BR>> --Bandwidth and Colocation provided by Easynews.com --<BR>> <BR>> asterisk-video mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> <A href="http://lists.digium.com/mailman/listinfo/asterisk-video" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-video</A><BR><BR><BR>
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