similar to: Wideband (G722) MeetMe

Hi all, I?ve read that meetme works at G711 (ulaw), so asterisk would down-mix a number of parties using G722, is that still correct? If so, i?ve also read that Joshua Colp was/is working on a replacement (conf_bridge?) that works with G722. If this is this still in active development are there any planned timelines? If it?s in 1.6.0.6, and i?ve just missed it or it?s been renamed please be nice

Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is "g722/8000" which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp

In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions. Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be

Has anyone found a way to enable the g722 codec as a prefered codec in the Polycom provisioning files for the 550's? I couldn't find a pref for voice.codecPref.IP_550. What needs to be put into the allow field (sip.conf) for asterisk to allow the codec? -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz

Hello, I am working with several SIP projects that use g722, or are trying to do so, with pjsip library. According to pjsip team's interpretation of g722, it works with 14bits PCM for input/output, so pjsip basically 'converts' the audio sample from 16 bits to 14 when encoding and vice-versa. Some implementations don't do 16<->14 bits conversion, so when pjmedia talks to

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

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:

Tomorrow, Friday August 21st at noon EDT, we'll be on the rampage again with the VoIP Users Conference, much of which concerns Asterisk and the Asterisk community. We've covered a lot of ground in the past 2 1/2 years, and had a lot of great guest presenters. In the meantime, there's a new gadget to test at PhoneFromHere.com, you can hear the wideband demo on the ZipDX.com bridge:

Hi, The ITU has been churning out a series of codecs which stretch older codecs to support wideband or super wideband and stereo. Does anyone know of these things being used in the real world? They push the idea of these things offering high compatibility, as the narrowband bitstream is embedded in the wideband stream. However, I'm skeptical this offers any real world advantage.

What do you mean by "implement"? You're just using the Opus built-in PLC (passing NULL), right? The PLC generally attempts to find periodicity and replicate it. I guess if your signal isn't periodic it can lead to a repetition that isn't great. It's something that could probably be improved in the PLC. Cheers, Jean-Marc On 08/25/2015 01:21 PM, Scott Boekweg wrote:

Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? Thanks, Madhuri ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

I missed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa

Hi all, I would like to know if something I am trying to do is possible. I currently using 1.8 and would like to make recordings in wideband (16khz) I have an aastra 6739i which supports the g722 codec. I made an agi application in which I would like an user to record a promptlist. I use the agi command record_file. Which format setting do I need to be able to record in wideband?

BTW, when you have something working and stable, I could include it in the main Speex distribution. Jean-Marc Le mar 18/02/2003 à 20:57, Bernard Blackham a écrit : > On Tue, Feb 18, 2003 at 06:09:43PM -0500, Jean-Marc Valin wrote: > > Le mar 18/02/2003 ? 17:38, John Hayes a ?crit : > > > If I encode something in ultra-wideband, can I decode it in wideband by > >

> Hmmm, define working and stable :) By that I mean that you're fine with releasing it with your name on it and not be afraid to get flamed. > <braindump topic="speexcat"> > It began as a merge between speexdec and speexenc from 1.0beta3, > with the encoding/decoding removed, and simply piped in and out from > ogg streams. I never expected it would work joining

Hello, We currently have code for sampling at 8 KHz at device layer. In order to have minimal code change while still doing wideband I am thinking of packing two 20 ms frames of the low band data as input to Wideband mode for encoding. Is this a feasible idea? I actually tried it out. Speech came out at the decoding end, but I could occasionally hear low pitched, high frequency

Hi, I found this list of Speex bitrates in the mail archive. http://www.xiph.org/archives/speex-dev/200306/0004.html Can somebody confirm that this list is correct? I am wondering about the following: - On the Speex website it says: "Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps." while the bitrates listed here are e.g. 84400 for

Hi! In a message posted on May 31 Jean-Marc wrote in a topic about AEC: > - set sampling rate to 8 kHz (at least for now) Does it mean that echo cancellation algorithm won't work in wideband mode? I can't get aec to work well in my application and I am wondering whether it's not a matter of using wideband instead of (preferred?) narrowband mode? Thanks in advance for any