similar to: enabling G.729 for SIP

Hello Jean-Marc: I've been watching the speex development from its inception because I and several Tech Startup Connection members have a very important application for this voice encoding/decoding. Further, we are quite familiar with ACELP as implemented in G.729. As far as I know Speex is also ACELP ... yes/no? Question ... what do you see as the advantages of ACELP compared to MP3? I

> I've been watching the speex development from its inception because I > and several Tech Startup Connection members have a very important > application for this voice encoding/decoding. Further, we are quite > familiar with ACELP as implemented in G.729. As far as I know Speex > is also ACELP ... yes/no? No. Speex is CELP (Code-excited linear prediction), but not ACELP

Thank you Jean-Marc. My understanding is that G.729 is a telephone codec, so there must have been some reason why its developers went to 10ms/frame. Do you know why that might be? From a recent post on this list I saw somebody talking about your decoded sample rate being 8KHZ/sec. and then he mentioned that being 160 bytes at 20ms/frame. That said, I take it that your decoded samples are

Thank you. You're right ... my error ... I meant to say 12 bytes (including the 2 bytes for VAD). And it is 10ms/frame. No matter ... thank you for the SPEEX specs. In terms of quality, what SPEEX bit rate compares with G.729 at 8kbps data rate please? Is there some reason why you chose the 20ms frame rate? Do you keep that same frame rate for the different bit rates? The faster frame

im using g729, but the bw usage is ~26 kbps per call, my gateways (cisco) support g723 and the bw between the gateways is ~18 kbps per call. Much better than the ~62 kbps of the g711. if you plan to be a voip provider you "must" go with compression codecs, especially if you want your customers to browse the internet while having a call. i.e. : We give voip phones (grandstream) to our

I have two ATA186 running 2.14 and 2.15. I see in the SIP debugs that both ATAs keep on sending SIP registration packets over and over. The flow is as follows: Asterisk receives REGISTER packet Asterisk sends 100 trying, 200 ok with an expire of 3600. Kurt __________________________________ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time.

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, Does the G.729 module support adding more licences??

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

I thought I'd post my experiences for the benefit of anyone else who may be at the point I was when I first started with asterisk. I have 2 incoming analog lines (north eastern U.S., Verizon) where one is set to ring if the first is busy. I bought a bare-bones system from abs-pc with the following components: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard)

Clicking on the "For more information, click here" link on the Digium site nice brings back up the same page I was looking at before, without any additional G.729 information that I can see. I'm wondering if some kind asterisker out there could provide us neophytes with some "typical scenarios" where that codec would be useful to us. For instance, I assume that it

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

Hi all, Has anyone tried conveying signaling tones (DTMF) through speex? If so, can i get an idea of the lowest bit-rate at which I can do so with 'tolerable distortion'. G.728 does so and G.729E (11.8 Kbps mode) is 'claimed' to do so but even if true, G.729 (rather G.729I - 11.8/8/6.4 kbps) does not allow me to go below 6.4 kbps if needed. Basically I am looking for a single codec

I have done my homework on this, I hope. I have a customer with an ATA186 who uses Nufone as his IAX provider. His network operations center in the Bahamas was destroyed by the hurricanes, and I'm helping him rebuild. We have a nagging problem getting his ATAs (located in public IP space) to talk through his IAX provider (Nufone) to the outside world. As far as we know, things worked OK

Hi Jean, Thats great news for me to start off with as I was planning to go with 16 Kbps ADPCM keeping in mind the issues and options I had. Now, whether the additional computation cost is worth the significant bandwidth savings, I have to see. Just wondering if it is possible to extend this logic to G3 and G4 fax as well, i.e. using a higher bit-rate and complexity mode for modem or fax instead of

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

In case a system is incapable of fax relay or if it is disabled, one of the easiest and safest options is to go for 40 kbps ADPCM compression (for fax upto 14.4 kbps)..even am new to this problem and the fair bit of seraching which i've done seems to suggest that the standard sloutions are to simply 'bypass' it else compress using ADPCM (40 k for fax upto 14.4 k, 32 k for fax upto 9.6