RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
One of our Telephony Server 5000 modules will throughput between 2,000 and 2,500 SIP calls with streams if it is doing no other work. One of these days we will again announce the details of the ongoing benchmarks that we perform with the help of system engineers from a major computer manufacturer. The key statement is if it is doing no other work. If a server is playing IVR or hosting conferences, throughput declines in unpredictable ways depending on the actual mix of work. So when we spec a system for a particular call volume we use relatively conservative engineering to ensure that the system can handle the peak load. In real applications, we rate a box at less than half of its peak call throughput. So for 5,000 calls, we'd probably use five servers plus an extra one for failover. Someone trying to do that same amount of work with PC servers might need up to four dozen of them in a complex configuration with a central voicemail store. The load balancing and system management problems are considerable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, February 02, 2006 9:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion I don't think they are doing it with one Asterisk box. They did say one rack of servers. Well, that might mean up to 50 computers if they are using blade servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Thursday, February 02, 2006 10:21 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion [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,
RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
There you go. if it is doing no other work is key phrase. A lot of PC can do that these days if all it has to do is re-route packets to different destinations, and guess what, if you make sure silence compression is turned on at the endpoints, you can claim even more streams can be passed through. The trict here is how * stores the mapping pair and how effiecent its lookup process is. I have not looked at this part of the code in *, but would be interesting to find out. On another topic. How many calls do you think one server can handle if every calls goes to a different IVR script of its own? Lets assume there is no trans-coding. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William Boehlke Sent: Friday, February 03, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion One of our Telephony Server 5000 modules will throughput between 2,000 and 2,500 SIP calls with streams if it is doing no other work. One of these days we will again announce the details of the ongoing benchmarks that we perform with the help of system engineers from a major computer manufacturer. The key statement is if it is doing no other work. If a server is playing IVR or hosting conferences, throughput declines in unpredictable ways depending on the actual mix of work. So when we spec a system for a particular call volume we use relatively conservative engineering to ensure that the system can handle the peak load. In real applications, we rate a box at less than half of its peak call throughput. So for 5,000 calls, we'd probably use five servers plus an extra one for failover. Someone trying to do that same amount of work with PC servers might need up to four dozen of them in a complex configuration with a central voicemail store. The load balancing and system management problems are considerable. ___ --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
RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
We run the same benchmarks on PC servers and get a small fraction of that throughput with streams. If you're doing better, that's great. The facile response is how much money do you have since it's a tuning issue. For example, you get more throughput if you hold the IVR in RAM. But, as you know, there are no simple answers. If the IVRs are short, say ten seconds, you'll be limited by call setup times instead of call throughput and could peak at 500 calls or even fewer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Friday, February 03, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion There you go. if it is doing no other work is key phrase. A lot of PC can do that these days if all it has to do is re-route packets to different destinations, and guess what, if you make sure silence compression is turned on at the endpoints, you can claim even more streams can be passed through. The trict here is how * stores the mapping pair and how effiecent its lookup process is. I have not looked at this part of the code in *, but would be interesting to find out. On another topic. How many calls do you think one server can handle if every calls goes to a different IVR script of its own? Lets assume there is no trans-coding. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of William Boehlke Sent: Friday, February 03, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion One of our Telephony Server 5000 modules will throughput between 2,000 and 2,500 SIP calls with streams if it is doing no other work. One of these days we will again announce the details of the ongoing benchmarks that we perform with the help of system engineers from a major computer manufacturer. The key statement is if it is doing no other work. If a server is playing IVR or hosting conferences, throughput declines in unpredictable ways depending on the actual mix of work. So when we spec a system for a particular call volume we use relatively conservative engineering to ensure that the system can handle the peak load. In real applications, we rate a box at less than half of its peak call throughput. So for 5,000 calls, we'd probably use five servers plus an extra one for failover. Someone trying to do that same amount of work with PC servers might need up to four dozen of them in a complex configuration with a central voicemail store. The load balancing and system management problems are considerable. ___ --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
RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
I don't think they are doing it with one Asterisk box. They did say one rack of servers. Well, that might mean up to 50 computers if they are using blade servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Thursday, February 02, 2006 10:21 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] RE: 5,000 concurrent calls system rolloutquestion [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: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com 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
RE: [Asterisk-Users] RE: 5, 000 concurrent calls system rolloutquestion
On Fri, 3 Feb 2006, Wai Wu wrote: I don't think they are doing it with one Asterisk box. They did say one rack of servers. Well, that might mean up to 50 computers if they are using blade servers. 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. ^^^ If you look at the datasheet http://www.signate.com/pdf/TelephonyServer.pdf its pretty clear its a cluster. I dont think anyone would be able to route 5,000 RTP streams on a single CPU these days, no matter how studly it is. -Dan ___ --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
