--On February 3, 2006 3:56:21 AM +0900 Vic <[EMAIL PROTECTED]> wrote:
Hi, Joash,
thank you for your email. I was very relieved to hear that someone was
already doing this.
Can you please tell me more about your test? Why did you test it in a
first place?
For me, we need to come up with a system that needs to:
1. Handle 5,000 inbound SIP calls
2. offer IVR capability
3. Billing
You'd probably have to do some of your own work on this. * makes 'CDR'
records but...well...you have to be careful how you do your scripts if you
want legible/useable CDRs. There are some apps out there though that will
process and do some sort of billing for CDRs not sure of what where.
I thought that Asterisk would be up to the task, but, I am not sure as
to:
1. How many servers should I consider? 4? 10? Obviously, we will be
talking about probably core Xeon servers if this is what we need.
I'd say atleast 10....maybe more...depending wholly on codec/transcoding
and amount of IVR scripting.
2. How hard would it be to implement?
Well...since your not well versed with *, and you're having trouble
understanding the difference between a protocol and a codec, it might be
really difficult for you. You might want to farm it out. There are a LOT
of * consultancies out there now. If you can get up to speed on asterisk
pretty quickly and the various protocols and codecs then it's not
impossible. The kicker is all the management/maintenance UI's and such.
But you might be able to use something like Signates sigMAN (never used it
or their products).
3. How bad is g729 quality?
4. IVR : if the call is SIP, can we do prompts without transcoding?
You're confusing protocols with codec's here again. SIP is not a codec.
That said if your SIP client is using GSM and there are GSM prompts
available then the asterisk playback functions will use the GSM encoded
prompts.
Earlier you'd mentioned using POTS lines coming in/out. If you're
gatewaying 5k POTS lines you'll need a lot of machines. Because you'll be
doing a lot of transcoding POTS is ulaw or alaw (depending on where in the
world you are) and unless you use (uncompressed) ulaw or alaw on your SIP
clients (very unlikely scenario) you'll be transcoding to/from GSM. G.729,
or whatever you're using.
Any other suggestions that you might have would really be appreciated.
Joash Herbrink <[EMAIL PROTECTED]> wrote:
I have tested an asterisk server with over 5000 concurrent calls.
The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet
connection on a cisco 3560 switch.
This works, but puts some serious stresses on the system.
Why don't u considered using g.729 codec, this will at least lower the
bandwidth consumption significantly, and, you can overcome the CPU
resource issue by just using a server grade multi CPU xeon server.
I would never the less still connect the system via 2 ethernet
connections, just for some redundancy, as mentioned before in this
thread.
Bandwidth should be about 24 kbps (half duplex) per call
So, 5000 * 24 is roughly 120 mbps, so a gigabit Ethernet should do just
fine.
Joash
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dustin
Wildes
Sent: Wednesday, February 01, 2006 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout
question
Dinesh Nair wrote:
On 02/01/06 09:29 Damon Estep said the following:
Ok, now lets go for 5000 of them. 160kbps*5000=800000kbps or 800mbps -
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at 80%
utilization and not at least start to fall apart?
additionally, 5000 simultaneous SIP calls at 20ms intervals will send,
5,000 * 50 * 2 = 500,000 packets per second (full duplex).
not too many boxes can handle such packet load, in spite of the
relatively small packet sizes.
Why not bond multiple NICs together to do a load balance output? Would
provide redundancy as well.
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