similar to: How to config speex?

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

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

Hello, I am interested to use Speex but need to know If I can use it under our project's constraint. We need to use a low cost micro controller on a very small PCB. This means there's no space for an external DSP unit. Can someone tell me if using an AVR32 (Atmel uc) which has a floating point calculation unit and DSP instruction set and running at ~50Mhz for a narrow band

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

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

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

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

I want to choose one codec between iLBC and Speex,because the codec is run on the embedded system,so the performance is the key factor to be encode and decode to be a real system. so ,i like to ask which codec is better in perfomance on the same hardware system? thank you -------------- next part -------------- An HTML attachment was scrubbed... URL:

> Obviously this is Jean-Marc's decision and I'm not telling > him not to support this compiler. I am however pointing > out that this compiler is yet more work for very little > payoff. In the case of my project, the proponent of C++ > Builder sent me a huge, monsterously ugly and totally > unmaintainable patch to add C++ Builder support. Needless > to say, that

Hi, I have set up the asterisk in Fedora 7 with SJphone, SJphone is successfully registered under asterisk console.....so i had some tests on the extension......but there were messages saying that something(probably .gsm files) were being played, but i heard nothing.....there were also some warning messages saying that in Asterisk CLI ERROR:sound device currently not available(some sort like

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

blink : It use iLBC also. Voice over IP RTP: A Transport Protocol for Real-Time Applications RFC3550 RCTP: Real Time Control Protocol attribute in Session Description Protocol RFC3605 SRTP: The Secure Real-time Transport Protocol RFC3711 DTMF: Dual-tone Multi-frequency Signaling RFC2833 and inband MWI: Message Summary Event Package RFC3842 Speex and G722: Wide-band Internet Codecs G711, iLBC

Hi Who can give me list of companies or VoIP product strongly based on speex? I want to know whether it is good enough for real enviroment like packet losing etc.. Regards bay

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

Hi Is it possible to use only the speex echo cancellation module w/o using the speex codec? Here's the scenario: 1. I have my voice recorded in PCM audio file format 2. I want to cleanup the recorded voice by removing any echo included in the audio file/buffer 3. can I just use the ff APIs? - SpeexEchoState* speex_echo_state_init() - int speex_echo_ctl() - void speex_echo_capture()

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

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I