So let's assume I am going to use G.729A. I am looking at using
Polycom IP601 phones which support G729A directly, so the only licenses
I believe I would need are for the calls going to voicemail or in the
menu system at once - realistically that number never exceeds 5
simultaneous, since the phones can handle the CODEC and no transcoding
is needed, so those do not need licenses according to
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing.
It looks to me like, for testing, I can get a couple of the polycom
phones and have a server using an IP on the unused T1.
Assuming that is correct (which I will write up as an article for the
Wiki if anyone is interested when this is all done), the next thing I
need is a provider of VoIP service. Also, it seems like the server
would go on the outside of my firewall with holes punched through for
the phones which would be on the ind=side of the firewall. Would that
be correct?
W
Steve Langstaff wrote:
Remember to add the RTP, UDP and IP overheads.
And then just do the math.
Depends on the codec. If you are using ulaw, you will only be
able to have about 23 calls. If you use g729 you can have as many as
187 simultanious calls on a data T1.
Remember, you have 1544Kbs of bandwidth.
g279=8Kbs per call
uLaw=64Kbs per call
Just do the math.
bp
On 6/19/06, Warren <[EMAIL PROTECTED]>
wrote:
Steve,
I want to end up with a system that will let me send and receive voice
calls. I guess what I want to do depends on the best way to do that.
Can I do more than 23 (decent sounding) voice calls on a data T-1 with
someone else handling the final part of the call to the copper for me?
If so than that is my likely final destination.
I have a channelized voice T-1 currently plugged into my meridian
system, but I would like (if realistically possible) to do as much of
this over IP as possible for maximum flexibility. Is that a pipe dream
or just silly given the current state of technology?
I am lucky enough to work for a company that is letting me take my time
with this, test the various options and come up with the proper
solution. I am assuming (I know: dumb to assume) at this point that
VoIP over a T-1 to a provider that can then route it to hard phones for
me would be the way to go. Similarly, I would point my 800 number to a
DiD hosted by a VoIP provider that would then route the call back to
me. If that is an incorrect assumption, please let me know.
Regards,
Warren
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
