similar to: Comparison

Le vendredi 10 juin 2005 ? 21:27 -0400, SteveK a ?crit : > 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

i have been working on a voip client that goes head-to-head with skype in technological terms. for this, we used speex wide-band codec. without the denoiser or the pre-processor, i find that speex quality at 16 khz sampling, 16-bit samples (mono) to be clearly superior to anything that skype offers. even though, at the moment, i am not using packet loss compensation, i find that speex is

Hi, First, you can see a comparison of the codec features at http://www.speex.org/comparison.html As for quality/bitrate, the first thing is that Speex supports a lot more settings (from 4 to 42 kbps) and does wideband (16 kHz sampling), which iLBC doesn't do. I've only tested iLBC once, but I've found that Speex has a better quality for the same bit-rate (or lower bit-rate for the

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

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

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

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Hi One of my softphones is supposed to support g711 , however I am getting these errors and a 404 not found when I try to make a call from it. However on xlite it works fine using g711. Below is the log of the phone that is not working. Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0

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! everybody: We used opus-codec for a VOIP gateway. The GW is running at a UBUNTU server. The opus stream is transcoded to G711 pcmu stream.So there are many opus codecs running simultaneously. We noticed that if there more than 5 streams in. the voice then has notisable glitchs.More streams in, worse voice got. Then we write test code for opus-codec which encode a .pcm file simultaneously.

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hey all, So we have been advised from this thread https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 to not use http put as it is not in real-time, instead they are suggesting the use of SDP, is that something that icecast supports? Or does anyone have other ideas on this? ~stephen On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: > Hi, > > On 11

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi, On 11 May 2013 15:32, Stephen Mahood <mv at cyberunions.org> wrote: > Thank you for your interest in this, you description is as accurate as I > can see. > >> From my perspective your challenges will be to get the containers right. >> WebM for audio+video >> Ogg for audio >> >> Also (I'm not that familiar with webRTC) you might need to reencode

Hi Stephen, > So we have been advised from this thread > https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 > to not use http put as it is not in real-time, instead they are > suggesting the use of SDP, is that something that icecast supports? Or > does anyone have other ideas on this? The imminent Airtime 2.4.0 release has support for Opus, and it

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

comment inline. On Wed, 16 May 2007, Jean-Marc Valin wrote: >> 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