similar to: super high bandwidth codec

Hi all, I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? Which licence could I use and how many ones are required (only one per phone or also for voicemail and MOH)? Regards Amaury -------------- next part -------------- An HTML

I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXXXXXX, while the UK has: 441xxx

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of

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Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

Hi, I'd be interested in comments from any users of the vonage service in the UK? http://www.vonage.co.uk is the website. Where are the servers located, traceroute would be useful. What is the general reliability like? Thanks Mike

Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354

Hi, Is there any comparison made between Speex and iLBC free codec? How would they compare in terms of quality, bitrate and CPU utilization? Thanks, Joe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20050610/b79a3f46/attachment.htm

I'm not an expert either, but I see people choosing iLBC over speex all the time with asterisk; partly it's because they have more market share in hardphones, and partly it's because of marketing and such. (another reason is that iLBC source is included in asterisk, and speex is only compiled in if you have the speex development stuff on your machine when you compile

I already have ilbc installed on my system. The files are: /usr/include/ilbc/iLBC_decode.h /usr/include/ilbc/iLBC_define.h /usr/include/ilbc/iLBC_encode.h /usr/lib/libilbc.a /usr/lib/libilbc.la /usr/lib/libilbc.so -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0 -> libilbc.so.0.0.0 /usr/lib/libilbc.so.0.0.0 However, if I do a "make" in asterisk-1.4.19, it will not detect that libilbc.a

Hi kind developers, I need select soon the best freeware VOIP codec, I see that all competitors are using iLBC because of the separate packets management. How speex behave in case of packets drop? Why other choice all iLBC? Thank you for any kind answer. Best regards. ------------------------------------- Roberto Della Pasqua Http: www.dellapasqua.com Email/Msn: roberto@dellapasqua.com

I'm being PHBed into a VOIP project, and Speex sprang to mind. Bandwidth is going to be a fairly serious issue for us. With regards to a Speex enc/decoder, I was wondering: Rick Kane and David Siebert have already asked about this, but seem to have gotten very different responses - the former a call to arms, and the latter a "well, if you do it, it'll get done." What's the

I'm a new Asterisk user. I've managed to set it up to do everything I want except sound good. Currently, Asterisk sounds considerably worse than my cell phone. I know VOIP can be _better_ than my cell phone, because I've heard Skype do it. (Using 32k iLBC, I believe.) I did an experiment with audio quality: 1) I made a recording which was pretty good. I used an iSight

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

>> The main idea is that Speex supports many bit-rates, but for one reason >> or another, some modes may be left out in implementations (e.g. for RAM >> or network reasons). What we're saying here is that you should make an >> effoft to at least support (and offer) the 8 kbps mode to maximise >> compatibility. > > I understood this. But as you may know: the

Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the

You definitely need to have separate decoders for separate streams. It has been mentioned before that inter-frame state is critical to achieving the level of quality for bandwidth that Speex offers. This differentiates it from iLBC, a codec whose claim to fame is that it treats each frame independently. I'm not sure what's hard about maintaining multiple decoder states, unless you are

This is what I found useful when put samba and CUPS together. diff -ur samba-3.0.1.orig/docs/htmldocs/CUPS-printing.html samba-3.0.1/docs/htmldocs/CUPS-printing.html --- samba-3.0.1.orig/docs/htmldocs/CUPS-printing.html 2004-01-28 10:56:08.000000000 +0800 +++ samba-3.0.1/docs/htmldocs/CUPS-printing.html 2004-01-28 11:23:07.000000000 +0800 @@ -1850,7 +1850,11 @@ parameter (which tries to prepare

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out