Displaying 13 results from an estimated 13 matches for "voiperized".
2005 Oct 11
3
R: echo cancellation
On Tue, 11 Oct 2005 10:36:51 +0200, Jean-Marc Valin
<Jean-Marc.Valin@USherbrooke.ca> wrote:
> Source code at:
> http://people.xiph.org/~jm/speexclient/
I rewritten my program so it is more similar to yours, I grabbed your code
for storing and retrieving echo, and I have better results, I think the
echo is cancelled in about 50% but it still can be heared. I think there
is a
2009 Jan 13
9
FWD and Asterisk
I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID , I dont see any call landing in
asterisk.
In extension.conf (vicidial) file I have
exten => 2062036895 ,1,Ringing()
exten => 2062036895 ,2,Wait(1)
exten => 2062036895 ,3,Answer()
exten => 2062036895
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2005 Apr 04
1
Feedback
Concerning the VAD in SPEEX, I've been using "VoIPerized" a phone using
SPEEX sampling at 16K, I use a Plantronics USB headset. I find that I have
to set the VAD quite sensitive in order not to lose "S's" from the start of
a word/sentence. the result is that I'm often sending packets
unintentionally resulting in packet loss from t...
2005 May 08
1
speex 1.1.8
Hi,
I saw there is a new release of speex.
What is this SPEEX_PLC_TUNING option for?
And is there a more complete list of changes? Because I like to determine if it is interesting to update to 1.1.8
Greetings Jeroen de Kleijn (developer of VoIPerized)
2005 Oct 11
0
R: echo cancellation
Hi,
I implemented the echo cancellation from speex in my Windows VOIP-client VoIPerized --> www.voiperized.com
Greetings Jeroen de Kleijn
-----Original Message-----
From: speex-dev-bounces@xiph.org on behalf of hs
Sent: Tue 10/11/2005 4:02 PM
To: speex-dev@xiph.org
Cc:
Subject: Re: R: [Speex-dev] echo cancellation
On Tue, 11 Oct 2005 10:36:51 ?, Jean-Marc Valin...
2005 Apr 18
3
speex voice seems to be bit breaking over long distance.
...here
got any source code to reference from?
Regards,
YueWeng
--- "Kleijn, J.S. de" <J.S.d.Kleijn@student.tue.nl>
wrote:
> Hi,
>
> You should implement a jitter buffer, this buffers
> the audio to prevent the 'breaking'.
>
> My VoIP application called VoIPerized has a jitter
> buffer. You can try it to see if you then don't
> experience the breaking of the voice.
>
> Greetings Jeroen de Kleijn
>
> -----Oorspronkelijk bericht-----
> Van: speex-dev-bounces@xiph.org namens Tay YueWeng
> Verzonden: ma 18-4-2005 11:28
>...
2020 Jan 29
0
Invitation for OpenSIPS Summit 2020 Call for Paper
Hello fellows VOIPer,
If you want to share with the rest of the VoIP & RTC community some
news, interesting or breaking through ideas, or even more, some
experience you had in terms of designing, integrating or operating
various solutions or platform based on Open Source Softwares, then you
should consider submitting a paper for the OpenSIPS Summit 2020 in May,
Amsterdam.
2004 Sep 07
1
interpolation of lost frames
...rmfull for the audioquality that at the end of the talkspurt one frame is interpolated? Or would this be almost inpossible to hear since the last audiopacket in the talkspurt already is about silent? This obviously depends on how speex interpolates a frame.
Greetings Jeroen de Kleijn
Developer of VoIPerized
2005 Apr 18
0
speex voice seems to be bit breaking over long distance.
...ds,
> YueWeng
>
> --- "Kleijn, J.S. de" <J.S.d.Kleijn@student.tue.nl>
> wrote:
> > Hi,
> >
> > You should implement a jitter buffer, this buffers
> > the audio to prevent the 'breaking'.
> >
> > My VoIP application called VoIPerized has a jitter
> > buffer. You can try it to see if you then don't
> > experience the breaking of the voice.
> >
> > Greetings Jeroen de Kleijn
> >
> > -----Oorspronkelijk bericht-----
> > Van: speex-dev-bounces@xiph.org namens Tay YueWeng
> >...
2009 Jan 13
1
FWD and IPCall
I tried this
http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html
But I am NOT getting call in asterisk.
SIP.conf file :
_________________
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
externhost=59.160.44.21
localnet=192.168.0.2/255.255.255.0
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test at 10.10.10.16:5060
;
2003 Oct 13
6
Asterisk Manager
Hello all,
Can I execute linux command like(ls, mkdir) through the Manager interface?
I can't seem to access the manual at digium.com. I keep getting 'Forbidden'
error. Looks like they are upgrading or something.
CF
2006 Jan 30
3
adress book
Hello to all
Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know
the best way of implement a centralized address book system.
Maybe the solution is LDAP, but these clients doesnt seem to support
LDAP.Who should contact the LDAP directory? the SIP clients or the SIP
server?
Thanks
Joao Pereira