[top-posting continued due to formatting sloth on my part]
So, then let me follow up with a few more comments:
1) I will make some assumptions from your note:
a) Asterisk is currently capable (unless something has "broken"
recently) of handling 2500 SIP-SIP calls with no transcoding,
including RTP sessions, if on an operating system and hardware that
is appropriately configured. This puts to rest some who have claimed
that 5000 channels is "impossible" with Asterisk regardless of
platform, at least according to Signate.
b) It is unclear if other channel drivers (IAX and Zaptel,
specifically) have had any testing with significant numbers of
channels.
c) It is unclear if anything other than pure RTP passthrough is
viable in these configurations. Maybe IVR causes collapse. ?
2) Still no claims or comments on the specific testing methods, or on
methodology. I'm left still scratching my head as to if this is
actually possible, since there is no specific claim that can be
verified. While I hope that your system can do those numbers (it
would help me greatly in the future!) I can't say that I'm confident
yet. I'll follow up in private email for further discussion.
3) Nobody else has thus far taken the bait and made any comments
about their systems. I appreciate Signate's comments; they seem to be
the only ones to publicly claim large-scale throughput using Asterisk
in a public forum. Most other people who claim thousands or even
high hundreds of connections do so offhand, without responding to
second questions when I raise my figurative eyebrows.
4) There are still no notes on other problems with scale here. I've
had systems with several hundred simultaneous SIP connections, but
"sip show channels" sure does start to take a while. What _other_
problems crop up, but don't necessarily cause a "failure" condition?
5) I will agree that most SIP testing systems are currently too
pricey. I would love to find a well-connected network that rents out
a few of the better-known SIP testing tools to beat on Asterisk
installations in remote places for short periods of time. But this
has always been the case... test gear is a small market, and
expensive. Just look at the MSRP of new high-end HP Oscilloscopes if
you want to get a picture of price-gouging.
JT
At 11:21 AM -0800 2/2/06, William Boehlke wrote:
Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony
Server 5000. The throughput has little to do with Asterisk and a lot to do
with hardware design and operating system tuning. Our very minor code
changes were returned to the project last year.
The benchmark we used to make that initial claim was flawed, however we have
since replicated the throughput in a different way to save our marketing
bacon.
How we actually achieve the throughput is our intellectual property but we
have a number of customers who are scaling towards and past that traffic
level. One of these days we hope to be able to justify the very large fee
Hammer wants to extract from us to produce a third party verification.
In production environments, of course, systems do more than switch calls. We
think high volume system design using 32-bit systems of any kind is complex,
and it's difficult to replicate the volumes without actual customer traffic
- and by then it's too late. Where do you put voicemail? Where does the IVR
reside?
When someone needs to switch 5,000 calls with Class 5 services we would
specify a rack of servers. The good news is that it is one rack, not three
of them, but we need more than Asterisk alone, great though it is, to make
everything work.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, February 01, 2006 9:33 PM
To: [email protected]; [email protected]
Subject: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout
question
Signate sells a single server that can get you to the call volumes you
need.
Paul Mahler
<mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED]
www.signate.com
[snip]
Past conversations on this topic have generated quite a bit of
controversy within the Asterisk development community, both publicly
here on the list forums as well as in quite a few more quiet
discussions with people who often do not post but have extensive
operational experiences with Asterisk (most of whom monitor the -dev
list and whose replies will be suited to that audience.)
The subject of load on a single chassis is still the most contentious
issue to date. The Signate numbers of >5000 calls per chassis with
RTP are impressive, and there are others who claim more vaguely of
1000, 2000, or more calls into a single P4 server (with or without
media.) Others say that there are inherent limits in the Asterisk
code which prevent more than ~500 calls from being processed with RTP
at any one time. Opterons, FreeBSD, custom Linux loads, Solaris, and
other operating systems or hardware have been offered as the magic
bullets to increase call volumes. Who knows? (1) I will say that
extraordinary claims demand extraordinary evidence, which has been
pretty thin. I believe that most large call processing facilities
still run on distributed systems of some type, as was described in
the primary thread of this discussion on -users. (2)
I know that there are some projects towards testing Asterisk more
rigorously to determine these numbers. However, I would suggest that
the community at large could benefit from a more open examination of
high-end system claims immediately than these (better) long-term
tests which are progressing slowly (if at all.) Let's just look at
the "maximum" numbers. Running a big system? Selling a big system?
Tell us about it, in detail. What are the limits that have been hit?
Be specific. I keep seeing hand-waving, but no programmers have come
forward to say "It won't work because of the way X is implemented in
the file blah.c or libFOO."
To make a bad analogy: I don't want to see the street rods; I just
want to see the top-fuel, rocket-powered dragsters on the line. Any
takers? It sounds like Signate has a contender, but quite a few
people have said that it's impossible without serious modifications
to the code. Others have claimed (publicly or privately) that they
can match those numbers on different hardware.
Here are the criteria:
- Any O/S
- An unmodified version of Asterisk from SVN (or CVS)
OR patches must be available for inspection, as per the GPL
OR you must be a Digium license-holder (patches can be secret)
- All calls are IAX2 or SIP (both in and out)
- No transcoding of any type is required
- All calls are G.711, 20ms OR 30ms packet size
Documentation:
- All O/S documentation, kernel tricks, modules, hacks, patches, or
configuration elements should be documented, but proprietary
information need not be divulged if that is deemed "secret"
- Testing method must be reasonably documented
- Dialplans must be included
- SIP.conf files must be included
- All hardware must be fully described (part numbers required)
TEST #1:
All media must be handled by the server. This is for both legs of
the call. The "canreinvite=no" for SIP and "notransfer=yes" in IAX2
must be set for all calls.
TEST #2:
Media may or may not be handled by the server. Native transfers
should be allowed in both IAX2 and/or SIP.
(1) I have heard various people saying that it is "impossible" for
Asterisk to handle a large number of calls due to architectural
issues (no, it's not just from the people that you'd "expect" to hear
this from.) I've not been able to validate this one way or the other
recently. I am interested to hear what the developer community has
as a comment on this topic. I have an Empirix Hammer system at my
company, but honestly I just don't have the time to set it up to do
testing due to day job time constraints...
(2) There are so many ways to spread calls across an Asterisk array
it makes my head spin, but the question STILL comes down to "how many
calls can a single chassis handle?" Even in a farm of servers, there
has to be a numerator in that ratio.
JT
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