RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Signate runs asterisk on a SGI box. Nothing special, do yourself a favor and just buy the SGI box yourself. In fact I have 3 SGI boxes for sale. Ill rip off the Signate labels and sell them to you. I worked out an asterisk load balance solution, so I dont need one all powerful PC. I distribute the load to many PCs... Doug From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Thursday, February 02, 2006 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, several of your mentioned signant as a viable option. Has anyone ever used them? Are there any reviews for their products? Did they just put together a lot of Asterisks into a large scale PC? (I am still struggling with the concept) Thanks, Vic ___ --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] 5,000 concurrent calls system rollout question
On 02/02/06 06:13 [EMAIL PROTECTED] said the following: On Wed, 1 Feb 2006, Kristian Larsson wrote: Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. Not to nitpick, but freebsd has routed 1M+pps using commodity hardware. thanx, i wanted to point this out but didnt want to inadverntly start a linux vs freebsd flame war. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] 5,000 concurrent calls system rollout question
On Thu, 2 Feb 2006, Dinesh Nair wrote: On 02/02/06 06:13 [EMAIL PROTECTED] said the following: On Wed, 1 Feb 2006, Kristian Larsson wrote: Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. Not to nitpick, but freebsd has routed 1M+pps using commodity hardware. thanx, i wanted to point this out but didnt want to inadverntly start a linux vs freebsd flame war. 1Mpps is no longer only the realm of 'big iron'. linux can do it on commodity hardware too. there's no magic in 1Mpps anymore. of course thats just routing the packets. actually doing something with the contents is a different matter entirely. i doubt theres any hardware which can handle 5,000 simultaneous voip calls on a single box. -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
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
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 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. 2. How hard would it be to implement? 3. How bad is g729 quality? 4. IVR : if the call is SIP, can we do prompts without transcoding? 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=80kbps 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. ___ --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 ___ --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
[Asterisk-Users] 5,000 concurrent calls system rollout question
Hi, several of your mentioned signant as a viable option. Has anyone ever used them? Are there any reviews for their products? Did they just put together a lot of Asterisks into a large scale PC? (I am still struggling with the concept) Thanks, Vic ___ --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] 5,000 concurrent calls system rollout question
--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 10maybe 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=80kbps 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. ___ --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 -- Genius might be described as a supreme capacity for getting its possessors into trouble of all kinds. -- Samuel Butler ___ --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] 5,000 concurrent calls system rollout question
--On February 3, 2006 4:07:05 AM +0900 Vic [EMAIL PROTECTED] wrote: Hi, several of your mentioned signant as a viable option. Has anyone ever used them? Are there any reviews for their products? Did they just put together a lot of Asterisks into a large scale PC? (I am still struggling with the concept) Well I've nebver used it but any single box solution is going to have to have custom hardware and some custom code in asterisk or asterisk channel module to run it. A PC can't do echo cancellation on 5k channels. Can't do codec on 5k channels. It might be able to do (light/simple/short) IVR on 5k channels though. Thanks, Vic -- Genius might be described as a supreme capacity for getting its possessors into trouble of all kinds. -- Samuel Butler ___ --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] 5,000 concurrent calls system rollout question
On Tue, Jan 31, 2006 at 06:29:07PM -0700, Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? No no no. First you come to the conclusion that you have 800Mbps of traffic, but this is bi directional thus 400Mbps in each direction. Then you're comparing you're 800Mbps to 1Gbps. If you compare bi directional you need to count the card as 2Gbps. So you are nowhere near 80% but closer to 40%. To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] 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=80kbps 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. ___ --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] 5,000 concurrent calls system rollout question
On Wed, Feb 01, 2006 at 03:38:21PM +0800, Dinesh Nair wrote: On 02/01/06 09:29 Damon Estep said the following: Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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. Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. FreeBSD is using a Radix tree for routing lookups, by using Linux you may choose something better performing such as LC-trie where you're able to push quite a lot more. But this is pure routing done in the kernel, with asterisk you have to bring the packets to userspace and back limiting the performance by quite a lot. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] 5,000 concurrent calls system rollout question
On 02/01/06 15:54 Dustin Wildes said the following: Why not bond multiple NICs together to do a load balance output? Would provide redundancy as well. the issue here would be the increased interrupts needed to handle the load, not necessarily a bandwidth related issue. using device polling (available in freebsd) could mitigate this somewhat, but i'm not yet sure what it adds to latency and jitter of the voice conversation. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] 5,000 concurrent calls system rollout question
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=80kbps 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. ___ --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] 5,000 concurrent calls system rollout question
Even if you could, you wouldn't want to use just one system to handle this call load. What happens when you lose a power supply or a hard drive, or any other random failure? I would think you would want a more robust design. While you can go the signate way and use SGI hardware to increase your load per footprint, you can alos go the way of a large cluster of low priced systems as well. I would do something like this: Two SIP router systems (all signalling, no media) that all SIP devices (end user UA's provider trunks etc.) communicate with in a load balanced fashion. These two routers recieve registrations all SIP signalling. They keep track of dynamic UA locations (SER or Asterisk could be used here). They use a SIP 302 redirect where possible and re-invite where redirect isn't supported to route call requests to a cluster of asterisk systems. For 5000 calls with no media, two systems should be good enough for N+1 redundancy (in other words one server is enough, but you have two so you can fail one at any time). Behind this you stick as many asterisk servers as is needed based upon the hardware and it's load ability. Again, N+1 should be your minimum design basis for the number of systems. The two routing systems should have a method of knowing the load on each node so that when redirecting a call, they can do so intelligently. This would also allow you to build in the ability to take nodes offline for maintenance or other requirements. Just throwing together a bunch of asterisk systems and using 'round-robin' routing will quickly become a management nightmare. While this can definately be done using asterisk, like someone else said, if you want to do it right, you are going to be looking at the need for a strong implementation team. -Chris ___ --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] 5,000 concurrent calls system rollout question
I guess I just assumed that that the connection to asterisk would have to be IP since it is absolutely impossible to connect ~208 T1s directly to a single asterisk server. You would have to use an external media gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :) Not necessarily. Granted that you will not be able to have have that many T1's on one system but if the load is spread across multiple Asterisk Media converters you should be able to do anything and scale your system much better. Lets consider for example the following scenario: -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- || --- | Main * | | server | | | --- || -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- This will let you serve 192 channels per media server. Media servers will only need to convert PRI-IP so a cheap DIY Dual Core Xeon MP with 4MB cache would be more then enough to process/compress 196 channels in/out of 2 TE406P's. Also media servers do not need much RAM, hard drives and can run from flash cards. My preference would be convert all the traffic coming out of media servers to G.729 and IAX2 trunk it to main server. IAX2 trunking will save you MANY interrupts and will improve your bandwidth utilisation between Media and Main servers. With this setup you can run Media and Main servers on private gigabit network which would be more then enough to handle IAX2 trunked G.729 traffic from media servers. Network redundancy can easily be achieved between Media and Main servers by adding NIC's to each and using many known techniques (bonding, routing, VRRP, etc, etc). The Main Asterisk server can be setup with load balancing/failover. Media servers will need to be aware of this. The good thing in the setup like this is that its easy to scale up when needed, you're not exposed of loosing all of your T1's if one of media servers fail, you can easily add more T1's in your setup. The Main server would need a quad gigabit card (intel is a good choice) and since it would not be hampered by Zaptel traffic and it would not need to do any transcoding (except for odd voicemail usage, that could be send to another server) you could use 2xDual Core Xeons. A separate dual port (for redundancy) gigabit card would be used to serve SIP clients. We're working with one of the ISP's on testing and perhaps implementing this setup for them. This setup is considerably cheaper then $1M proposed Cisco setup and can be made as reliable as Cisco solution is. Please don't get me wrong, if I'd have $1M-$5M to spare would go for Cisco. But most of us don't have that much money and if we would, we would never be reading any messages on asterisk-users. Asterisk can be made as reliable and scales as good if not better then any Cisco solution and the fraction of the cost. Now imagine all of this with the new DS3000P in media servers! All hail Asterisk! :) ___ --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] 5,000 concurrent calls system rollout question
Still, Everybody is using T1 / E1 interfaces in servers. I would go for purpose build voice gateways. Vegastream or cisco GW are able to handle multiple T1/E1 connections easily. (make sure that in a cisco GW you get enough DSP capacity) In this scenario the asterisk server is just used to make sure everybody gets connected to each other. I a + 5000 call setup I would say some money is available to buy the dedicated voice gateways. I would go for the vegastream 400 series. Use 2 of them to prevent than all your PSTN connections are terminated in one machine. These kind of setups work with my customers (not 5000 concurrent calls) but we do have connected over 3000 phones to the Asterisk server in some locations. You can then use a good management GUI like scopserv (be sure to mention my name :-) :-) ) (www.scopserv.com) to manage you're asterisk servers in a easy way (and also keep configs synchronized) You will still be a hell of a lot cheaper then a cisco callmanager setup, and get far more performance and features then cisco will ever be able to offer you. Joash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Wednesday, February 01, 2006 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question I guess I just assumed that that the connection to asterisk would have to be IP since it is absolutely impossible to connect ~208 T1s directly to a single asterisk server. You would have to use an external media gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :) Not necessarily. Granted that you will not be able to have have that many T1's on one system but if the load is spread across multiple Asterisk Media converters you should be able to do anything and scale your system much better. Lets consider for example the following scenario: -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- || --- | Main * | | server | | | --- || -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- This will let you serve 192 channels per media server. Media servers will only need to convert PRI-IP so a cheap DIY Dual Core Xeon MP with 4MB cache would be more then enough to process/compress 196 channels in/out of 2 TE406P's. Also media servers do not need much RAM, hard drives and can run from flash cards. My preference would be convert all the traffic coming out of media servers to G.729 and IAX2 trunk it to main server. IAX2 trunking will save you MANY interrupts and will improve your bandwidth utilisation between Media and Main servers. With this setup you can run Media and Main servers on private gigabit network which would be more then enough to handle IAX2 trunked G.729 traffic from media servers. Network redundancy can easily be achieved between Media and Main servers by adding NIC's to each and using many known techniques (bonding, routing, VRRP, etc, etc). The Main Asterisk server can be setup with load balancing/failover. Media servers will need to be aware of this. The good thing in the setup like this is that its easy to scale up when needed, you're not exposed of loosing all of your T1's if one of media servers fail, you can easily add more T1's in your setup. The Main server would need a quad gigabit card (intel is a good choice) and since it would not be hampered by Zaptel traffic and it would not need to do any transcoding (except for odd voicemail usage, that could be send to another server) you could use 2xDual Core Xeons. A separate dual port (for redundancy) gigabit card would be used to serve SIP clients. We're working with one of the ISP's on testing and perhaps implementing this setup for them. This setup is considerably cheaper then $1M proposed Cisco setup and can be made as reliable as Cisco solution is. Please don't get me wrong, if I'd have $1M-$5M to spare would go for Cisco. But most of us don't have that much money and if we would, we would never be reading any messages on asterisk-users. Asterisk can be made as reliable and scales as good if not better then any Cisco solution and the fraction of the cost. Now imagine all of this with the new DS3000P in media servers! All hail Asterisk! :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Still, Everybody is using T1 / E1 interfaces in servers. I would go for purpose build voice gateways. Vegastream or cisco GW are able to handle multiple T1/E1 connections easily. (make sure that in a cisco GW you get enough DSP capacity) In this scenario the asterisk server is just used to make sure everybody gets connected to each other. I a + 5000 call setup I would say some money is available to buy the dedicated voice gateways. I would go for the vegastream 400 series. Use 2 of them to prevent than all your PSTN connections are terminated in one machine. These kind of setups work with my customers (not 5000 concurrent calls) but we do have connected over 3000 phones to the Asterisk server in some locations. You can then use a good management GUI like scopserv (be sure to mention my name :-) :-) ) (www.scopserv.com) to manage you're asterisk servers in a easy way (and also keep configs synchronized) You will still be a hell of a lot cheaper then a cisco callmanager setup, and get far more performance and features then cisco will ever be able to offer you. Joash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Wednesday, February 01, 2006 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question I guess I just assumed that that the connection to asterisk would have to be IP since it is absolutely impossible to connect ~208 T1s directly to a single asterisk server. You would have to use an external media gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :) Not necessarily. Granted that you will not be able to have have that many T1's on one system but if the load is spread across multiple Asterisk Media converters you should be able to do anything and scale your system much better. Lets consider for example the following scenario: -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- || --- | Main * | | server | | | --- || -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- This will let you serve 192 channels per media server. Media servers will only need to convert PRI-IP so a cheap DIY Dual Core Xeon MP with 4MB cache would be more then enough to process/compress 196 channels in/out of 2 TE406P's. Also media servers do not need much RAM, hard drives and can run from flash cards. My preference would be convert all the traffic coming out of media servers to G.729 and IAX2 trunk it to main server. IAX2 trunking will save you MANY interrupts and will improve your bandwidth utilisation between Media and Main servers. With this setup you can run Media and Main servers on private gigabit network which would be more then enough to handle IAX2 trunked G.729 traffic from media servers. Network redundancy can easily be achieved between Media and Main servers by adding NIC's to each and using many known techniques (bonding, routing, VRRP, etc, etc). The Main Asterisk server can be setup with load balancing/failover. Media servers will need to be aware of this. The good thing in the setup like this is that its easy to scale up when needed, you're not exposed of loosing all of your T1's if one of media servers fail, you can easily add more T1's in your setup. The Main server would need a quad gigabit card (intel is a good choice) and since it would not be hampered by Zaptel traffic and it would not need to do any transcoding (except for odd voicemail usage, that could be send to another server) you could use 2xDual Core Xeons. A separate dual port (for redundancy) gigabit card would be used to serve SIP clients. We're working with one of the ISP's on testing and perhaps implementing this setup for them. This setup is considerably cheaper then $1M proposed Cisco setup and can be made as reliable as Cisco solution is. Please don't get me wrong, if I'd have $1M-$5M to spare would go for Cisco. But most of us don't have that much money and if we would, we would never be reading any messages on asterisk-users. Asterisk can be made as reliable and scales as good if not better then any Cisco solution and the fraction of the cost. Now imagine all of this with the new DS3000P in media servers! All hail Asterisk! :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Based on our benchmarking, I am VERY skeptical of this number. Im guessing that you dont really have RTP streams going through the NIC. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joash Herbrink Sent: Wednesday, February 01, 2006 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question 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=80kbps 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. ___ --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] 5,000 concurrent calls system rollout question
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kristian Larsson Sent: Wednesday, February 01, 2006 2:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question On Tue, Jan 31, 2006 at 06:29:07PM -0700, Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? No no no. First you come to the conclusion that you have 800Mbps of traffic, but this is bi directional thus 400Mbps in each direction. Then you're comparing you're 800Mbps to 1Gbps. If you compare bi directional you need to count the card as 2Gbps. So you are nowhere near 80% but closer to 40%. Not true, each g.711 call generates 1x 80kps FULL DUPLEX RTP stream, if the call originates g.711, enters *, gets bridged to another channel, and goes back out to the phone as an additional g.711 call you are generating 160 kbps of FULL DUPLEX traffic per call. 160kbps full duplex * 5000 calls IS 800mbps FULL DUPLEX! Until someone steps up and says we are currently doing this in a PRODUCTION environment and it has been working for X months I will stand by my opinion that a single chassis 5000 call implementation of * with RTP streams bridged is IMPOSSIBLE for many reasons, this being one of them, the interrupt servicing being the next, the cpu load being 3rd, and so on... And if it can be done, I would still not do it - too many eggs in one box. This should be done on 10x high end servers, 500 calls each, no transcoding, round robin call origination - and even at that - 500 calls per box will require a 2 to 4 CPU box with GigE NICs Holy cow - just writing 5000 CDRs in a short time interval will bring a box to its knees. There have been many theoretical reasons why it can't be done posted (including mine), and no real world experience posted. There are reasons for that. To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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] 5,000 concurrent calls system rollout question
Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo ==+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=== == Don't forget to account for the fact that the calls would be bridged, so each call is really 2! = 10,000 RTP streams. ___ --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] 5,000 concurrent calls system rollout question
I guess I just assumed that that the connection to asterisk would have to be IP since it is absolutely impossible to connect ~208 T1s directly to a single asterisk server. You would have to use an external media gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :) Not necessarily. Yes! Granted that you will not be able to have have that many T1's on one system but if the load is spread across multiple Asterisk Media converters you should be able to do anything and scale your system much better. Lets consider for example the following scenario: -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- || --- | Main * | | server | | | --- || -- -- | * Media | | * Media | |server | |server | | 2x TE406P| | 2x TE406P| -- -- The main sever is still connected via IP, correct? Does not matter if you use * for media gateways or an APX8000 - the only trunking options to get to the main box are IP based. This will let you serve 192 channels per media server. Media servers will only need to convert PRI-IP so a cheap DIY Dual Core Xeon MP with 4MB cache would be more then enough to process/compress 196 channels in/out of 2 TE406P's. Also media servers do not need much RAM, hard drives and can run from flash cards. My preference would be convert all the traffic coming out of media servers to G.729 and IAX2 trunk it to main server. IAX2 trunking will save you MANY interrupts and will improve your bandwidth utilisation between Media and Main servers. With this setup you can run Media and Main servers on private gigabit network which would be more then enough to handle IAX2 trunked G.729 traffic from media servers. Network redundancy can easily be achieved between Media and Main servers by adding NIC's to each and using many known techniques (bonding, routing, VRRP, etc, etc). The Main Asterisk server can be setup with load balancing/failover. Media servers will need to be aware of this. The good thing in the setup like this is that its easy to scale up when needed, you're not exposed of loosing all of your T1's if one of media servers fail, you can easily add more T1's in your setup. The Main server would need a quad gigabit card (intel is a good choice) and since it would not be hampered by Zaptel traffic and it would not need to do any transcoding (except for odd voicemail usage, that could be send to another server) you could use 2xDual Core Xeons. A separate dual port (for redundancy) gigabit card would be used to serve SIP clients. We're working with one of the ISP's on testing and perhaps implementing this setup for them. This setup is considerably cheaper then $1M proposed Cisco setup and can be made as reliable as Cisco solution is. Please don't get me wrong, if I'd have $1M-$5M to spare would go for Cisco. But most of us don't have that much money and if we would, we would never be reading any messages on asterisk-users. Asterisk can be made as reliable and scales as good if not better then any Cisco solution and the fraction of the cost. Now imagine all of this with the new DS3000P in media servers! All hail Asterisk! :) ___ --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] 5,000 concurrent calls system rollout question
On Wed, 1 Feb 2006, Kristian Larsson wrote: Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. Not to nitpick, but freebsd has routed 1M+pps using commodity hardware. Its tricky but can be done. -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
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
The main sever is still connected via IP, correct? Does not matter if you use * for media gateways or an APX8000 - the only trunking options to get to the main box are IP based. Are seriously going to tell me that a quad xeon/opteron would not handle traffic from 4xGIG cards?? :) ___ --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] 5,000 concurrent calls system rollout question
I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 On 1/29/06, Wai Wu [EMAIL PROTECTED] wrote: To handle 5000 calls coming in over a PRI, you'd need 210 or so T1s or 170 E1's.All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) [Wai Wu] He not talking about PRI here, but rather SIP to SIP There is no way possible that you're going to pump that amount of data through a PC. Don't care about codecs and dialplans, PC's just don't have that sort of internal bandwidth from peripherals. If all the endpints support reinvite and he is not doing any voice processing at all, there is hardly any data going through the PC ___ --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] 5,000 concurrent calls system rollout question
--On February 2, 2006 9:55:15 AM +1100 Boris Bakchiev [EMAIL PROTECTED] wrote: The main sever is still connected via IP, correct? Does not matter if you use * for media gateways or an APX8000 - the only trunking options to get to the main box are IP based. Are seriously going to tell me that a quad xeon/opteron would not handle traffic from 4xGIG cards?? :) Probably not no. Especially not if you're not using special IRQ load balancing software. Xeon especially not since all teh CPUs share a FSB. The opteron *might* if the motherboard were designed such that the multiple pci busses terminated at multiple procs, and each proc had local memory attached to it. Then you might be able to approach that rate. ___ --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] 5,000 concurrent calls system rollout question
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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] 5,000 concurrent calls system rollout question
In our experience, it's not a bandwidth limitation. If you do nothing special, interrupt servicing for a single NIC on our high throughput hardware maxes at something in excess of 1,000 calls when you are keeping the streams. I don't believe you can get even that far on a PC server, but we haven't tried. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, January 31, 2006 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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] 5,000 concurrent calls system rollout question
I saw your previous post that signate has a solution - care to elobaorate? (invitation for a shameless plug) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Boehlke Sent: Tuesday, January 31, 2006 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question In our experience, it's not a bandwidth limitation. If you do nothing special, interrupt servicing for a single NIC on our high throughput hardware maxes at something in excess of 1,000 calls when you are keeping the streams. I don't believe you can get even that far on a PC server, but we haven't tried. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, January 31, 2006 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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 ___ --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] 5,000 concurrent calls system rollout question
On 1/31/06, Damon Estep [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; No it's not, the math was meant for the T1 calculations, reread the post I replied to. First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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] 5,000 concurrent calls system rollout question
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 9:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question On 1/31/06, Damon Estep [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; No it's not, the math was meant for the T1 calculations, reread the post I replied to. I guess I just assumed that that the connection to asterisk would have to be IP since it is absolutely impossible to connect ~208 T1s directly to a single asterisk server. You would have to use an external media gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :) First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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? To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --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] 5,000 concurrent calls system rollout question
On 02/01/06 09:29 Damon Estep said the following: Ok, now lets go for 5000 of them. 160kbps*5000=80kbps 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. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] 5,000 concurrent calls system rollout question
On Sun, Jan 29, 2006 at 05:24:12PM +1000, Rob Thomas wrote: The question is somewhat ludicrous, and I'm slightly surprised that no-one has sat down and done the maths about bandwidth utilization. So I did. To handle 5000 calls coming in over a PRI, you'd need 210 or so T1s or 170 E1's. Yes. All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) No. Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds Using g729 it's a lot less. And even if it was 320Megabytes/s that equals a little over 3Gbps not 32. There is no way possible that you're going to pump that amount of data through a PC. Don't care about codecs and dialplans, PC's just don't have that sort of internal bandwidth from peripherals. Well, with your above miscalculations, no. But 3.2Gbps is possible with a few boxes. And there are PCs with at least four different PCI-X buses, that's 4GB/s so it might even be possible with one machine. Anyway, it's still a lot of calls. If you do, honestly, need to handle 5k calls, you'd probably have to have a bank of Cisco 5850s doing the termination - With a max of 5 DS3 (1 DS3 = 28 T1's) into each one, you'll need 8, or probably 9 as you'd want to have one as a hot spare. Each of those DS3's would go into some beefy switching fabric (note, that each one is producting 225mbit) and you'd have some sort of asterisk box with huge internal bandwidth handling each one. Cross connect all 9 asterisk boxes via 10Gbit networks (note, you'll need PCI-16x 10g cards) and have a pair of voicemail servers. I'd suggest a pair of big Sun boxes. Then, of course, you have the issue of getting the calls _out_ of the asterisk machines. You've just doubled your bandwidth requirements, so you'll need to double up on the asterisk machines, and split the network up further. I'd take a guess that you could do it under USD$1million (just for hardware) but I wouldn't be surprised if it was USD$10million. I'm happy to sell you any of this 8-) --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Sunday, 29 January 2006 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, can we make it a bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! --- - ___ --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 ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Jan ___ --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] 5,000 concurrent calls system rollout question
On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] 5,000 concurrent calls system rollout question
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet. check this out. http://web1.egvrn.net/tokata/VoIP%20Bandwidth%20Consumption.pdf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kristian Larsson Sent: Monday, January 30, 2006 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] 5,000 concurrent calls system rollout question
5k+ simultaneous calls (in/out) are becoming normal with the kind of call centers being opened in my country during the past 24 months (Panama, Central America). Take Dell Corp. for example. the call center they have here is about 3k people taking/making calls (internal, to/from US, Europe, Asia). Other Call Centers are in that figure too. For me, this thread seems a good learning point to calculate how to do that with asterisk. Thanks to the people who answered here. On 1/30/06, Kristian Larsson [EMAIL PROTECTED] wrote: On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] 5,000 concurrent calls system rollout question
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet. And what is 'one voice packet'? Jan ___ --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] 5,000 concurrent calls system rollout question
One voice packet is about 20ms of sampling depending on the codec. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, January 30, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question 64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet. And what is 'one voice packet'? Jan ___ --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] 5,000 concurrent calls system rollout question
Vic a écrit : Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. What exactly do you do with these calls? We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. If you are just handling SIP signaling and routing a solution such as SIP Express Router is much more appropriate than Asterisk for this kind of volume. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. It depends on what you are trying to do. If you are transcoding 5,000 simultaneous calls, it's going to cost a lot of money, wether you use Asterisk or not... -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] 5,000 concurrent calls system rollout question
On Sun, 2006-01-29 at 17:24 +1000, Rob Thomas wrote: [snip] If you do, honestly, need to handle 5k calls, you’d probably have to have a bank of Cisco 5850s doing the termination Or have a look at the Lucent APX8100 box for some added carrier class humpf. Supports more than 8000 DS0's (channels) and does transcoding in hardware DSP's so well suited to handle your 5000 concurrent calls and you don't need a stack of them like with the Cisco 5850. Weblink: http://www.lucent.com/products/subcategory/0,,CTID+2013-STID+10443-LOCL +1,00.html Datasheet: http://www.lucent.com/livelink/090094038000eb91_Brochure_datasheet.pdf Like Rob I'd love to sell this to you but I doubt Lucent would even pick up the phone to answer my how to become a VAR enquiry. Best contact them directly :) Regards, Patrick (no affiliation with Lucent) ___ --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] 5,000 concurrent calls system rollout question
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Patrick Sent: Sunday, January 29, 2006 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question On Sun, 2006-01-29 at 17:24 +1000, Rob Thomas wrote: [snip] If you do, honestly, need to handle 5k calls, you'd probably have to have a bank of Cisco 5850s doing the termination Or have a look at the Lucent APX8100 box for some added carrier class humpf. Supports more than 8000 DS0's (channels) and does transcoding in hardware DSP's so well suited to handle your 5000 concurrent calls and you don't need a stack of them like with the Cisco 5850. Weblink: http://www.lucent.com/products/subcategory/0,,CTID+2013-STID+10443-LOCL +1,00.html Datasheet: http://www.lucent.com/livelink/090094038000eb91_Brochure_datasheet.pdf Like Rob I'd love to sell this to you but I doubt Lucent would even pick up the phone to answer my how to become a VAR enquiry. Best contact them directly :) Regards, Patrick (no affiliation with Lucent) The original poster of this message stated in an earlier message that the calls would be handed off to him SIP, so the media conversion is being done buy an upstream carrier, presumably on a Lucent or Sonus. With the growing availability of SIP origination and termination, high density channels banks like the APX8000 are becoming items only needed by wholesale carriers. Of course this varies by geographical region, but to use a APX 8000 you need at least PRI service over DS1/E1, and ideally PRI service over DS3/E3. The challenge I see with a 5000 INBOUND call setup originated SIP is that the calls will need to be load balanced across many * boxes, no 1 asterisk box is going to take 5000 CONCURRENT calls (500 would impress me). I would suggest; Check to see if the SIP origination provider can give you a round robin delivery of calls over 10 or so * boxes (IP addresses), or find an external method of doing it yourself (like a smart session border controller). IF the calls are terminated to hardphones or softphones (as opposed to purely IVR), make sure you can do RTP re-invites so, when appropriate, the RTP stream is offloaded from * (but consider the impact of doing so). Calculate bandwidth needs carefully - 5000 * 70-75kbps (a/ulaw plus packet overhead) requires a GIG-E IP link from you SIP provider and some very robust networking in between. Terminating 5000 calls on * is relatively uncharted ground, there MAY be some others doing it, but good luck getting them to reveal the company jewels. At the very least, this type of implementation would require a team of the VERY BEST asterisk consultants - might want to call Mark himself if you are serious. Damon ___ --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] 5,000 concurrent calls system rollout question
To handle 5000 calls coming in over a PRI, youd need 210 or so T1s or 170 E1s.All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec)[Wai Wu] He not talking about PRI here, but rather SIP to SIP There is no way possible that youre going to pump that amount of data through a PC. Dont care about codecs and dialplans, PCs just dont have that sort of internal bandwidth from peripherals. If all the endpints support reinvite and he is not doing any voice processing at all, there is hardly any data going through the PC ___ --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] 5,000 concurrent calls system rollout question
Have you verified that you are actually sending sound over the RTP streams? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith Sent: Friday, January 27, 2006 11:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question What's you mix of calls going SIP/IAXand to PSTN? We've done some benchmark experiments on a 3GHz HT box with 1GB of ram, mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls over 4 PRI spans. Its running MusicOnHold into 60 of the channels, playing various GSM prompts into the other 60. The user cpu usage is about 25%, the system cpu about 25% also. We can add to that 5000 registered SIP peers and 5000 registered IAX2 peers - total of about 100 registration refreshes per second. That adds about 40% more user CPU and pretty much fills up CPU. Audio quality is still perfectly fine, and PRI slips few and far between. Load average for the whole mix sits between 5 and 10. About 550Kb/s out and 400Kb/s out on the ethernet for the registration traffic. Also on www.voip-info.org - search for dimensioning Rob On 1/28/06, Vic [EMAIL PROTECTED] wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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] 5,000 concurrent calls system rollout question
Signate sells a single server that can get you to the call volumes you need. Paul Mahler [EMAIL PROTECTED] www.signate.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Saturday, January 28, 2006 7:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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 ___ --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] 5,000 concurrent calls system rollout question
It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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] 5,000 concurrent calls system rollout question
I have different need. In the same issue Vic presents. It's 3000 concurrent calls from PSTN (E1s) to Voip (gsm). And the other way around. 3000 Voip calls (SIP/H323 gsm) to PSTN. no voicemail, but the user may get 5 seconds of help prompts initially. Thanks, On 1/28/06, Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] 5,000 concurrent calls system rollout question
Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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 ___ --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] 5,000 concurrent calls system rollout question
On Jan 28, 2006, at 7:15 PM, Vic wrote: Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? You can also use it as and end use agent (ie an ATA or a phone). I am using an AG-168V which is a cheapo ATA that supports IAX directly. This is nice because it simplifies ports and firewall issues. In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, can we make it a bt more feasible number? Transcoding is a big consumer of CPU for sure. This has nothing to do with SIP however and is related to the CODEC you are using at the end of the line and in between. If all you calls are coming in and being delivered in the same format (ie g729), then you don't need to transcode anything, and the CPU load is much lighter. In fact you can setup asterisk to make a native bridge of these calls. Perhaps you could try building a testbed? That's what I would do. Good Luck, Marty Zoa [EMAIL PROTECTED]> wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: > Hi, > > we are currently considering different options for rolling out a large > scale IP PBX to handle around 3,000 + concurrent calls. > > Can this be done with Asterisk? Has it been done before? > > I really would like an input on this. > > Thanks! > > > >___ >--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 ___ --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] 5,000 concurrent calls system rollout question
The question is somewhat ludicrous, and Im slightly surprised that no-one has sat down and done the maths about bandwidth utilization. So I did. To handle 5000 calls coming in over a PRI, youd need 210 or so T1s or 170 E1s. All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) There is no way possible that youre going to pump that amount of data through a PC. Dont care about codecs and dialplans, PCs just dont have that sort of internal bandwidth from peripherals. If you do, honestly, need to handle 5k calls, youd probably have to have a bank of Cisco 5850s doing the termination With a max of 5 DS3 (1 DS3 = 28 T1s) into each one, youll need 8, or probably 9 as youd want to have one as a hot spare. Each of those DS3s would go into some beefy switching fabric (note, that each one is producting 225mbit) and youd have some sort of asterisk box with huge internal bandwidth handling each one. Cross connect all 9 asterisk boxes via 10Gbit networks (note, youll need PCI-16x 10g cards) and have a pair of voicemail servers. Id suggest a pair of big Sun boxes. Then, of course, you have the issue of getting the calls _out_ of the asterisk machines. Youve just doubled your bandwidth requirements, so youll need to double up on the asterisk machines, and split the network up further. Id take a guess that you could do it under USD$1million (just for hardware) but I wouldnt be surprised if it was USD$10million. Im happy to sell you any of this 8-) --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Sunday, 29 January 2006 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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 ___ --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
[Asterisk-Users] 5,000 concurrent calls system rollout question
Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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] 5,000 concurrent calls system rollout question
What's you mix of calls going SIP/IAXand to PSTN?We've done some benchmark experiments on a 3GHz HT box with 1GB of ram, mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls over 4 PRI spans. Its running MusicOnHold into 60 of the channels, playing various GSM prompts into the other 60. The user cpu usage is about 25%, the system cpu about 25% also. We can add to that 5000 registered SIP peers and 5000 registered IAX2 peers - total of about 100 registration refreshes per second. That adds about 40% more user CPU and pretty much fills up CPU. Audio quality is still perfectly fine, and PRI slips few and far between. Load average for the whole mix sits between 5 and 10. About 550Kb/s out and 400Kb/s out on the ethernet for the registration traffic. Also on www.voip-info.org - search for dimensioningRobOn 1/28/06, Vic [EMAIL PROTECTED] wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] 5,000 concurrent calls system rollout question
Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! Check out Signate Telephony Server, http://www.signate.com/pbx.php ___ --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
