a few quesitions about G729

[voipgkavek]

Hi, because of one sip provider we have, we might need to install g729 licenses in the server. I have a few questions.

1) is the best option digiums?
2) The install according to their README file seems simple, are there any gotchas?
3) We have a live PBX and a hot spare, not running, but ready to take over in case the live one has problems. Do I have to buy double the number of licenses even though I will never be using both machines?
4) How is licensing calculated? The sip provider will give me 20 channels. Do I have to get these 20 or 40 licenses? (20 for trunk + 20 more for the endpoints? I have aastra hardphones and zopier pro and eyebeam pro softphones)
5)I read something about using "passthrough" if my endpoints already handle the G729 codec...I believe that our aastra, zoiper and eyebeam all have g729. How does passthrough work?
thanks,
-gk

 

[atsak]

1. The only option is Digium. They have the patent.
2. The install is simple. Just follow the instructions.
3. Hmm, don't know ask Digium. I think so as the license is bound to the MAC address.
4. Should be 20, the endpoints typically have their own.
5. I have never heard of that.

 

[voipgkavek]

thanks atsak,
I just tried what i think is pass-through and it seems to work, i only tried 2 outgoing calls.
What I did was remove all codecs except g729 from eyebeam. Since the provider has a higher priority for g729 both the SIP provider and the endpoint agreed on G729 and I had nice 2 way audio. I assume That is pass-through since I have not purchased licenses yet.
The call that worked was a direct call..dial a number..other side answers.
The call that did not work was one that required to enter a PIN. I am assuming its cause eyebeam requested only g729 and since asterisk had to convert the PIN request audio to g729 and didnt have a license it just rejected the call. Its just my guess, not sure.
So if this indeed its how it works I assume I can buy just a few licenses since they are only required for the IVRs. Would a call on hold be occupying a license since its sending MOH? Would it release it once it bridges the trunk with the endpoint?
thanks,
-gk

 

[atsak]

The reason the PIN didn't work is because g.729 doesn't deliver terribly reliable DTMF sometimes. So beware. I don't know for sure about the pass through / number of licenses. Bit beyond my understanding of the codec.

 

[tm1000]

Pass-through means that instead of the audio flow looking like this:

phone1 <--> asterisk <--> phone2

It looks like this

phone1 <--> phone2

EG: The audio 'passes through' the asterisk server without getting 'processed' (as you can see it actually never passes through the server but this is the term)

It's one of the coolest things Freeswitch does as it means NO transcoding on the server and therefore you can have many, many, many phones connected at one time.

 

[voipgkavek]

Atsak, sorry, maybe I wasn't clear. When I said it didnt work, I meant PiaF just rejected the call. It said something like "this call not allowed here." I didnt even get to the point were it asks for DTMF.
-gk

 

[voipgkavek]

tm100, so freeswitch does it, but asterisk doesnt? Like I mentioned earlier, I did get a pure G729 call to work even though i dont have any g729 licenses. I am confused.
I kept thinking about this, about 50% of call will probably be on hold concurrently, that means I still need to buy a lot of licenses. Or is there a way around this? audio encoded in g729 maybe? does that exist?
-gk

 

[malcolmd]

See the directmedia and directrtpsetup options in sip.conf.

That aside, if you're going to do any conferencing, app_meetme or app_confbridge, or even mod_conference (on FreeSWITCH), you're going to be doing transcoding. No way around that.

A G.729 prompt set for the asterisk-core sounds is available. See http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-g729-current.tar.gz or just look in your Asterisk source build (make menuselect).

Digium also provides a prompt conversion tool here:
http://www.digium.com/en/products/ivr/audio-converter.php

Or, better, just buy a single-channel license and run:
asterisk -rx "convert /tmp/mysound.wav /tmp/mysound.g729"

Cheers.

 

[voipgkavek]

thanks for the info Malcolm. I will look into those options. I suppose i would have to create a script that converts my prompts since most are in spanish.
The online tool is great for the professionally made promts..just last night I had to use VLC to convert one. this seems simpler.
About the directmedia and directrtpsetup...i just read a bit about them, but didnt quite understand

-gk

 

[Tekmon]

Just blew over my head and took the cap off to. G.729 developed years ago is licensed, I get that. When one buys a phone it's paid for right? Or any G.729 capable equipment. One can't converse with a server unless you have an end point phone. Why would digium even need to be licensed? Signaling 101 here.

 

[malcolmd]

The license is permission to use the patented G.729 algorithm.

When you buy a phone that supports G.729, the phone's manufacturer has already paid those royalties - or has bought a DSP from a DSP vendor that's already paid it and passed it on.

But, that's *just* for the phone's transcoding of G.729 - you talk in analog, the phone encodes into G.729 and puts it on the wire and/or decodes from G.729 on the wire and puts it to the speaker.

That gets G.729 into Asterisk.

If Asterisk has to to anything to transcode that G.729 media into any other format (audiohook for recording, conferencing, you want your voicemail stored on disk in wav format, etc), then Asterisk needs to be licensed to do that transcoding, also. That's where the G.729 licenses that Digium offer come into play.

The short of "why do you need to be licensed," is because G.729's patented.

Cheers.

 

[eCase]

Correct G729 Installation Instructions

Hi Guys
I came across this thread because I recently reinstalled PIAF and was looking for all the various tips I had learned on the forum over the past five months. I needed to install my G729 codec again, and it made me remember that there is a typo in the install instructions provided by Digium - when I first installed the codec a few months ago, it literally took me hours and hours of guesswork and searching online for me to struggle through the instructions - all because of a simple typo. The advanced users probably won't need this tip, but for the rest of us:

When you buy 1 or more licenses of the G729 codec, you will receive various emails from Digium. One contains the install instructions and looks like this:

"Hello Digium Customer,

The following is your key for X channels of G.729 for Order: XXXXXXXX

Key : G729-XXXXXXXXXXXX

Instructions for using your key may be found at:

1. http://downloads.digium.com/pub/telephony/codec_g729/README
"
The info contained at the link is all correct except the following:
2.1) Download the benchg729 utility to the root home directory of your Asterisk server.
Command-line Example for 32-bit Linux:
# cd /root
# wget http://downloads.digium.com/pub/telephony/codec_g729/\benchg729/x86-32/benchg729-1.0.7-x86_32 -O benchg729

I suppose it should look like this:
wget http://downloads.digium.com/pub/telephony/codec_g729/\benchg729/x86-32/benchg729-1.0.8-x86_32 -O benchg729

Hope this helps!

 

[atsak]

Atsak, sorry, maybe I wasn't clear. When I said it didnt work, I meant PiaF just rejected the call. It said something like "this call not allowed here." I didnt even get to the point were it asks for DTMF.
-gk

Ah sorry - that usually means the g729 is either not allowed in the conifg for the extension or it's not correctly licensed. . .