search for: exosip

I am dying here with linphone (not sure if it is crap software or just me being an idiot) but out of the box debian installations of two linphones fail with a "Got SIP response 415 "Unsupported Media Type" back from 192.168.1.3" Can anybody recommend a particular SIP soft phone that broadly satisfies the following criteria? 1. Run on linux. 2. Simple to use and setup. 3. Is

...ist (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf draft... tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

...> I'm not able to understand completely why: anyway, it > now can be compiled without breaking anything. > > I'll now test wether the aec have an impact or not :) > tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > > On Mon, 6 Nov 2006, Aymeric Moizard wrote: > >> >> >> On Mon, 6 Nov 2006, Jean-Marc Valin wrote: >> >>> That's quite a strange problem. Considering that Speex doesn't have any >>> &...

...er you said that the SDP tells the mode information. Right. But a decoder like speex can always be configured from information contained in the incoming stream. That makes your application more interroperable. Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ > Is SDP what you called in-band signalling? > Lianghu > > On 12/6/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: >> >> In a "normal" frame that doesn't have in-band signalling (which nobody &...

...tically detect the mode (narrowband/wb..) > of the RTP stream I'm receiving. > > I guess this information is included in frames? Isn'it? > How can I read it? > > tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > _______________________________________________ > Speex-dev mailing list > Speex-dev@xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev > >

...speex to prove myself my >> application has a bug: if somebody has one, please send it. >> >> Sorry for bothering you with probably non-speex problems, >> >> Aymeric MOIZARD / ANTISIP >> amsip - http://www.antisip.com >> osip2 - http://www.osip.org >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ >> >> _______________________________________________ >> Speex-dev mailing list >> Speex-dev@xiph.org >> http://lists.xiph.org/mailman/listinfo/speex-dev >> >> > >

...e input has really low volume which is most > probably the main issue here. Still this is showing that > using the AGC potentially introduce DC component? Any > comment? > > Tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > _______________________________________________ > Speex-dev mailing list > Speex-dev at xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev > >

...d it? Just in case... I also wish to test another application using speex to prove myself my application has a bug: if somebody has one, please send it. Sorry for bothering you with probably non-speex problems, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

Hi, Speex-dev: ? I have some problems about complie speex on pxa-270. I use libspeex.so.1.4.0 at my platform after complie speex and all function is ok, but the cpu loading is 99%. How can i do to reduce the cpu loading? ? note: 1. My Platform are os: linux 2.6.19 , cc: gcc-2.95.3 , cpu: pxa-270, ogg: libogg-1.1.3, speex: speex-1.2beta3 2. I configure speex by the follow command. ./configure

Sorry, got the subject wrong in the last mail. My e-mail client is playing games with me...

...B While I should compile it with: -D_USRDLL I'm not able to understand completely why: anyway, it now can be compiled without breaking anything. I'll now test wether the aec have an impact or not :) tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ On Mon, 6 Nov 2006, Aymeric Moizard wrote: > > > On Mon, 6 Nov 2006, Jean-Marc Valin wrote: > >> That's quite a strange problem. Considering that Speex doesn't have any >> "init code" (stuff that executes wh...

Dear Sir/Miss: I'm very sorry to bother you,but I've found a bug in the Speex,Version 1.2beta1,and also I've have some questions. I'm deeply impressed by the work you have done in the coding and decoding of speech signal.But when using speex,I encoutered a bug,that is when in 32K mode,the window size is still 200 samples,but not 400 samples,the bug will bring the problem of

Aymeric Moizard a ?crit : > > I'm working on the echo canceller with the SVN, with Windows Mobile 5 > PocketPC. > > Obviously, I can't make it run correctly. What happens/doesn't happen? > My question: If I use the SPEEX_PREPROCESS_SET_ECHO_STATE, I have > changes in the sound: noise seems to be removed. Noise should be removed even without

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

...to re-adapt. So as soon as the (local) clocks aren't *perfectly* synchronised, the echo cancellation performance goes down to a point where it's mainly unusable. Jean-Marc > Tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > > On Wed, 1 Nov 2006, Tom Grandgent wrote: > >> Isn't this the same problem described starting at the bottom of >> this page? >> http://www.embeddedstar.com/articles/2003/7/article20030720-11.html >> >&...

Isn't this the same problem described starting at the bottom of this page? http://www.embeddedstar.com/articles/2003/7/article20030720-11.html Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > As it says in 5.4.1 of the good book "Using a different soundcard to do > > the capture and playback will *not* work, regardless of what you may > > think.

...s or delayed initiation (initial packets are often lost at the beginning on one side of the conversation). I think this approach will fit your need for keeping CELT as low as possible but mandatory for VoIP. tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

...are. It seems to be similar issue than the one described in your link, but the difference is really unpredictable and the resolution does not seems as simple... Anybody that wish to share experience on this? Tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ On Wed, 1 Nov 2006, Tom Grandgent wrote: > Isn't this the same problem described starting at the bottom of > this page? > http://www.embeddedstar.com/articles/2003/7/article20030720-11.html > > Jean-Marc Valin <jean-marc.valin@...

...) The only way would be to extend or reduce frames: so my question was: does anybody here have ever tried this in real time on audio streaming? Any simple idea to do this? I would love to see code on this... Tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

...est another application using speex to prove myself my > application has a bug: if somebody has one, please send it. > > Sorry for bothering you with probably non-speex problems, > > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > _______________________________________________ > Speex-dev mailing list > Speex-dev@xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev > >