similar to: G.729.1 - any interest?

1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet->PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

Hi, I just released unstable version 1.1.2 that contains more fixed-point work. Though it's still not 100% complete, enough have been done to make it run in real-time on ARM. In order to do that, compile with --enable-fixed-point --enable-arm-asm. All narrowband modes work in real-time with complexity 1 (some work with higher complexity) and some wideband modes also work (up to ~20 kbps) at

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent "generic" board out there that I could be investigating? It would be such a shame to waste a PCI slot that

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

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

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hi, I was wondering if anyone has experimented with Speex's wideband (16kHz) mode at lower quality settings. In particular I have been using quality 3, and with wideband input files the resultant frequency spectrum is limited to about an upper end around 3.5kHz (almost telephony quality bandwidth). Has anyone tried increasing the spectral bandwidth at the expense of lowering the

Hi all, I have recently added Speex support to my app (http://sbooth.org/Max/) and am in the process of creating a GUI for the user to control the codec parameters. I am new to Speex and as such I am not really sure which parameters are more important than others, and deserve prominent placement, etc. Is there a recommended GUI for people to follow or use as an example? Stephen

Kirk, Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you don't use one of these sample rates, you'll be messing up important assumptions deep within the codec. Why these sample rates? It's telecommunications tradition, rather than PC audio tradition. If you want an efficient and high quality format for voice chat, try 16kHz with VBR quality 6. You should see

I'm trying to get speex to encode a bit faster, mainly by rewriting a few functions in SSE and translating the GCC __asm__ to VC __asm. There's 2 functions I'm targeting, first is vq_nbest which consumes 40% of the time at high complexity and split_cb_search_shape_sign. Which consumes just over 30%. I've split out two functions from: cb_search_precompute_energy - loop at the

Jean-Marc Valin wrote: >> gcc version 3.4.5 (mingw special) > > Can you try 4.0 or 4.1 just to be sure? mingw (native gcc for win32) doesn't officially support using gcc 4.0 and 4.1 yet (apparantly there have been some issues), so there are no binary packages. But if you think it helps, I can compile gcc 4.1 and give it a shot. >> Compile flags: >> DEFINES +=

Hi all, If I use a SIP softphone and set to gsm codec clearly I get a 8K sample... if I change that to something like opus I get a much better sounding input... How do I get a "better" than 8K sampled input ? I "desire" to have that input be from a pipe. I have cd quality audio in this pipe - I would like to get that audio into asterisk to then send out to endpoints. How do I

Hi All We received following error .Please help us to sort out. WARNING[3281]: frame.c:1426 speex_samples: Had error while reading wideband frames for speex samples. Regards Nirukshitha ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search.

Compared to todays PDAs the GameBoy Advance (GBA) is low spec. it has a 16MHz ARM7tdmi RISC processor. 32-bit but it can handle 16-bit instructions in a mode that resembles "bytecode" (a 32-bit instruction is executed using a 16-bit instruction). I got a feeling that the GBA would need a FPU. For more information on the GBA visit this info page: