Hi, I am a begginer
i installed debian etch + asterisk 1.2 recently
i created 2 accounts
in my local LAN i have de asterisk box and an IPphone
when somebody from outside my LAN calls me, I hear my voice reaching the other side with a delay of 5 to 10 sec
i used a natted lan and a adsl internet conection
i installed a free g729 that i found on the web
what can be the problem?
thanks a lot
Best answer
Hi,
This answer is an extrapolation of http://forums.digium.com/viewtopic.php?p=63505#63505
--------------------------------------
[...]
I'd council them that what they're calling a "free" G.729 inplementation doesn't permit free use. It's code from Intel that provides the user the ability to use the code under an evaluation license, provided that user has registered for an evaluation license with Intel.
[...]
I'd also suggest to them that they attempt their call using a G.711 codec, u-law or a-law, for testing purposes. And, if they determine that they do need the bandwidth savings offered by the G.729 codec, that they investigate Digium's G.729 offering, for which Digium has secured the appropriate licensing from the G.729 consortium to sell a product.
If the call is an IP to IP call, and one user hears the sound of their own voice echoing back to them 5 seconds later, then they've got a very latent link and the IP phones are not doing a good job of echo cancellation. If it's not echo they're experiencing but instead delay as the audio goes from one phone to another, then there's simply too much latency. Perhaps the've got Asterisk installed on an underpowered machine.
--------------------------------------
From the same forum:
---------------------------------------------
Maybe it is for the g.729. It is delayed too much. You could try by g.711. if it is for the capablity of the pc. It is also possible for the network delay.
---------------------------------------------
AND:
---------------------------------------------
I would try ulaw. Outside of the legalities of the g729 codec is the fact that IMHO, it is crappy sounding.
In an Asterisk setup the Intel based G729 codec is kind of a wildcard, so at least for testing, try ulaw.
How does the outside caller connect to your LAN? Do you use a piece of hardware to terminate a PSTN line, or are you using an IP based phone company?
If you are using an IP based phone comapny, what are the ping times to them like? Are you using SIP or IAX? What kind of hardware do you have between the net and your asterisk box? Traceroute might be your friend.
How do calls between phones on your LAN sound? You can install a soft phone on a couple of PC's and see how that works.
--------------------------------------------------
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Dedicated server
Your best bet...
Submitted by admin on Tue, 2008-01-15 23:31.
Vote!Hi,
Your best bet is to ask this question to the Asterisk forums:
http://forums.digium.com
In order to post, you first have to register, here:
http://www.asterisk.org/user/register
I took the liberty to add your question to the forums:
http://forums.digium.com/viewtopic.php?p=63474#63474
Please follow it up.
Bye!
Merc.
Answer from the Asterisk forums
Submitted by Tony Mobily on Wed, 2008-01-16 22:53.
Vote!Hi,
This answer is an extrapolation of http://forums.digium.com/viewtopic.php?p=63505#63505
--------------------------------------
[...]
I'd council them that what they're calling a "free" G.729 inplementation doesn't permit free use. It's code from Intel that provides the user the ability to use the code under an evaluation license, provided that user has registered for an evaluation license with Intel.
[...]
I'd also suggest to them that they attempt their call using a G.711 codec, u-law or a-law, for testing purposes. And, if they determine that they do need the bandwidth savings offered by the G.729 codec, that they investigate Digium's G.729 offering, for which Digium has secured the appropriate licensing from the G.729 consortium to sell a product.
If the call is an IP to IP call, and one user hears the sound of their own voice echoing back to them 5 seconds later, then they've got a very latent link and the IP phones are not doing a good job of echo cancellation. If it's not echo they're experiencing but instead delay as the audio goes from one phone to another, then there's simply too much latency. Perhaps the've got Asterisk installed on an underpowered machine.
--------------------------------------
From the same forum:
---------------------------------------------
Maybe it is for the g.729. It is delayed too much. You could try by g.711. if it is for the capablity of the pc. It is also possible for the network delay.
---------------------------------------------
AND:
---------------------------------------------
I would try ulaw. Outside of the legalities of the g729 codec is the fact that IMHO, it is crappy sounding.
In an Asterisk setup the Intel based G729 codec is kind of a wildcard, so at least for testing, try ulaw.
How does the outside caller connect to your LAN? Do you use a piece of hardware to terminate a PSTN line, or are you using an IP based phone company?
If you are using an IP based phone comapny, what are the ping times to them like? Are you using SIP or IAX? What kind of hardware do you have between the net and your asterisk box? Traceroute might be your friend.
How do calls between phones on your LAN sound? You can install a soft phone on a couple of PC's and see how that works.
--------------------------------------------------