Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi, You might want to check out some of the EDUCAUSE http://www.educause.edu mailing-lists to find out what other universities are doing. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On 22 Nov 2008, at 00:06, Michael Collins wrote: Date: Fri, 21 Nov 2008 16:20:28 -0600 From: Terry Wilson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER in my case) was if I needed to load balance calls across multiple asterisk servers. If most of the phones are not separated by a NAT from Asterisk (as would be the case in something like a University network), the registration timeout could be set to a relatively high value w/o causing any problems which would cut down on some of the SIP traffic from registrations. In fact, I just ran some tests using SIPp and w/o any audio, using realtime w/ 10k accounts I can register 100/second while doing 10 calls/second. If you are looking just at registrations every 15 minutes or so, that is 90k devices that could register to asterisk. This was using 1.6.0.1 on my little HP amd64 development box--not anything near the kind of machine that you would probably install in a large installation. Asterisk just gets faster and faster. Some of the it isn't good at x stuff comes from experiences with older releases. In a HA and/or high volume scenario I worry about stuff like this that has been in tree since 1.0 or earlier and is in 1.6, channel.c lines 3825~3828: /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ That's not something I want in my high-end, high-capacity, high-availability production system! Actually that's exactly the kind of comment I _do_ want to see in an opensource platform. It is honest, open and an encouragement to others to think of a better fix. Discourage poor coding, critique the design etc - but please don't discourage this kind of commenting, it is the kind of thing that helps one find a bug _infinitely_ faster that you could without the clue the original author left for you. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On Fri, 21 Nov 2008, Al Baker wrote: Remember - You are going from a CARRIER GRADE purpose built piece of hardware with Software built under a rigid CMM with extensive soak-testing to software that has been developed under , shall we say, a somewhat less rigid and stringent methodology. You will be moving from an environment supported by hundreds of highly trained people, some with decades of TELCO experience to one where you support comes from a somewhat less seasoned group of individuals. 10,000 extensions ??? On Asterisk ??? You pays your money, you takes you chances. I know what a few friends who work/study in the astrophysics department of a university half an hour up the road from me would rather have - their new carrier grade switch built under a rigid CMM, etc. fails about once a month right now. Recently it was because it was start of term and it couldn't handle the additional call-load. They used to forward me emails from their support department as a bit of a joke, but they've stopped doing it now as it's way beyond a joke. They paid their money, took their chance with a full-commercial system and blew-it. I just wish I could get in there now, but it's too political a situation for an outsider to get anywhere. Whats equally annoying is that theirInnovations Centre (a sort of business incubator unit for graduates) has a very clever Asterisk system installed, but the main university seems oblivious to it all. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Strongly suggest to consider a Freeswitch/OpenSER implementation instead. Regarding purpose built and supported software.sometimes throwning billions of CMM software development to a product does not guarantee a good product... look at Micro$oft Vista. E http://Gpro.ws ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
2008/11/21 Yehavi Bourvine [EMAIL PROTECTED] Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello very interesting project you have, however asterisk is not a registry server, i suggest that you use opensips/opense/kamalio for your registrar, from where you dispatch to you asterisk servers, inside a good environment with a controlled network and nice tagged voip flow you could acheve a good results. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Thanks to everyone who replies so far! We have Nortel PBX'es with a support contract from one of the local VARs (Nortel does not give direct support here). In the last two weeks we had one of our exchanges down for three half days; one was after a failure, and the other two were when the technician came to fix remainders of the original problem and just did a 5 seconds restart which won't even cut calls. Yeh, the 5 seconds took 6-7 hours... BTW, they still do not know what was the original problem. So, why won't we save the big bucks we pay them, hire two professionals (who cost less) and support an open source code by ourselves? This way we depend on ourselves only. Thanks, __Yehavi: 2008/11/21 Grygoriy Dobrovolskyy [EMAIL PROTECTED] 2008/11/21 Yehavi Bourvine [EMAIL PROTECTED] Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello very interesting project you have, however asterisk is not a registry server, i suggest that you use opensips/opense/kamalio for your registrar, from where you dispatch to you asterisk servers, inside a good environment with a controlled network and nice tagged voip flow you could acheve a good results. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi Bourvine wrote: Thanks to everyone who replies so far! We have Nortel PBX'es with a support contract from one of the local VARs (Nortel does not give direct support here). In the last two weeks we had one of our exchanges down for three half days; one was after a failure, and the other two were when the technician came to fix remainders of the original problem and just did a 5 seconds restart which won't even cut calls. Yeh, the 5 seconds took 6-7 hours... BTW, they still do not know what was the original problem. Well, as everyone knows people don't generally have long memories, but I do. Back in the 1980s Northern Telecom installed a DMS-100 for Ameritech in Southfield, Michigan. Somewhere along the way the switching matrix was somehow undersized, fingers pointed both ways and heated words were exchanged between companies. Internally the switch was called Yo-Yo by the Ameritech employees. Just because you spent millions of dollars on a solution doesn't guarantee you five nines. Sometimes it costs millions more just to make it work right, both in fines and vendor fixes. Because of the Southfield fiasco when the largest exchange came up years later the contract went to ATT Bell Labs to install the largest 5ESS in the world in the Royal Oak exchange. Nortel wasn't even considered. Nortel also lost quite a bit of Ameritech business to the Siemens EWSD. Ameritech's Ohio Bell wouldn't even touch the DMS-100 for many years because of what happened in Southfield. This was at a time when all the switches needed to go digital to reduce power costs and comply with federal law, so there was a urgency to get hardware in place ASAP. Even with the pressure the DMS-100 was avoided. Due diligence is required on anything 10,000 people are going to be pounding on. Undersizing is common, and is only one of the roads that leads to Hell (I prefer Patterson Lake Road myself since I drive in from the North East). -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Al Baker wrote: Remember - You are going from a CARRIER GRADE purpose built piece of hardware with Software built under a rigid CMM with extensive soak-testing to software that has been developed under , shall we say, a somewhat less rigid and stringent methodology. You will be moving from an environment supported by hundreds of highly trained people, some with decades of TELCO experience to one where you support comes from a somewhat less seasoned group of individuals. But in choosing carrier grade (everyone calls their stuff that) vendors you are also going to a much smaller installed base and much lower total reporting and QA pool. I would take the sheer number and dynamism of the Asterisk installed base over their comparatively limited deployments, even if we grant the unsubstantiated premise that the latter is developed under a less rigid and stringent methodology. Let me put it this way: if I wrote a piece of software and sold it to 10 customers, it won't matter for overall product quality that I fix the problems they report and maintain it for them under the guidance of a rigid and stringent methodology. That's nice. Hope it fixes their problems. It is really of comparatively minor benefit to prospective future adopters. It's not nearly as valuable as simply doing the best I can with bug reports and test cases from hundreds of users. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Due diligence is required on anything 10,000 people are going to be pounding on. Undersizing is common, I think due diligence is THE key with any open source solution, including asterisk. I'll admit that I pretty badly screwed up one asterisk installation because I didn't adequately prepare it (shipped it to the customer and had their IT staff install - bad plan). And while I've never done a system anywhere near 10K extensions, I've had good experiences with some large-ish installations because I budgeted in the time for research and testing. I know that in the past there have been people on this list who have done very large scale asterisk deployments. Not sure if any of them are still around to comment. With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. and is only one of the roads that leads to Hell (I prefer Patterson Lake Road myself since I drive in from the North East). Hmm. You must live near Ann Arbor. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Noah Miller wrote: With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. Third. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED]wrote: [..snip..] With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. Is Asterisk even needed? - Gonzalo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Gonzalo Servat wrote: On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: [..snip..] With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. Is Asterisk even needed? Potentially, no. But if you intend to provide subscriber/PBX features, it is needed as a UA feature box(s). -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Noah Miller wrote: and is only one of the roads that leads to Hell (I prefer Patterson Lake Road myself since I drive in from the North East). Hmm. You must live near Ann Arbor. No, northern suburbs of Detroit. M-59 to US-23 S to M-36 W..To S. Howell St..Patterson Lake Rd..To Hell Ann Arbor is quite a bit South of Hell. Actually it's been some time since I've been to Hell but I'm sure it's frozen over today ;-) -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov [EMAIL PROTECTED]wrote: Gonzalo Servat wrote: On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: [..snip..] With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. Is Asterisk even needed? Potentially, no. But if you intend to provide subscriber/PBX features, it is needed as a UA feature box(s). And FreeSWITCH can't handle that? - Gonzalo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Is Asterisk even needed? Potentially, no. But if you intend to provide subscriber/PBX features, it is needed as a UA feature box(s). And FreeSWITCH can't handle that? Freeswitch can provide many PBX features with additional modules, but asterisk can provide more, and its implementations of such items are more time tested. One of freeswitch's big strengths is its ability to handle many SIP registrations. This is not asterisk's strength (at least not historically). One of Asterisk's big strengths is its multitude of services and features. This is not freeswitch's strength. Combine freeswitch and asterisk to get the best of both worlds. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Gonzalo Servat wrote: On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Gonzalo Servat wrote: On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: [..snip..] With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. Is Asterisk even needed? Potentially, no. But if you intend to provide subscriber/PBX features, it is needed as a UA feature box(s). And FreeSWITCH can't handle that? I suppsoe FreeSWITCH could, if you're so inclined. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Gonzalo Servat wrote: On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Gonzalo Servat wrote: On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: [..snip..] With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. Is Asterisk even needed? Potentially, no. But if you intend to provide subscriber/PBX features, it is needed as a UA feature box(s). And FreeSWITCH can't handle that? I suppose FreeSWITCH could, if you're so inclined. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On Fri, Nov 21, 2008 at 2:49 PM, Noah Miller [EMAIL PROTECTED]wrote: And FreeSWITCH can't handle that? Freeswitch can provide many PBX features with additional modules, but asterisk can provide more, and its implementations of such items are more time tested. One of freeswitch's big strengths is its ability to handle many SIP registrations. This is not asterisk's strength (at least not historically). One of Asterisk's big strengths is its multitude of services and features. This is not freeswitch's strength. Combine freeswitch and asterisk to get the best of both worlds. I was preparing a reply that would argue the need to have Asterisk involved if FreeSWITCH is there, but given the name of the list and the potential to piss people off, I'll leave it at that. - Gonzalo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
I know that in the past there have been people on this list who have done very large scale asterisk deployments. Not sure if any of them are still around to comment. With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Alex Balashov wrote: Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. Some of this is just a general design principle, nothing specific to registration. Once a VoIP platform gets to be big enough, a lot of the logical elements that go into it get centralised into distinct and dedicated components that are federated into a delivery platform. It's no longer considered a good idea at that point to have one process perform many different functions that have varying concurrency and blocking implications. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER in my case) was if I needed to load balance calls across multiple asterisk servers. If most of the phones are not separated by a NAT from Asterisk (as would be the case in something like a University network), the registration timeout could be set to a relatively high value w/o causing any problems which would cut down on some of the SIP traffic from registrations. In fact, I just ran some tests using SIPp and w/o any audio, using realtime w/ 10k accounts I can register 100/second while doing 10 calls/second. If you are looking just at registrations every 15 minutes or so, that is 90k devices that could register to asterisk. This was using 1.6.0.1 on my little HP amd64 development box--not anything near the kind of machine that you would probably install in a large installation. Asterisk just gets faster and faster. Some of the it isn't good at x stuff comes from experiences with older releases. If you are lucky enough to have a situation where you can re-invite media and keep it off of the asterisk box, it can handle huge loads. Terry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER Keep in mind this was in 1.4. They actually use hash lookups now anyway. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Date: Fri, 21 Nov 2008 16:20:28 -0600 From: Terry Wilson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER in my case) was if I needed to load balance calls across multiple asterisk servers. If most of the phones are not separated by a NAT from Asterisk (as would be the case in something like a University network), the registration timeout could be set to a relatively high value w/o causing any problems which would cut down on some of the SIP traffic from registrations. In fact, I just ran some tests using SIPp and w/o any audio, using realtime w/ 10k accounts I can register 100/second while doing 10 calls/second. If you are looking just at registrations every 15 minutes or so, that is 90k devices that could register to asterisk. This was using 1.6.0.1 on my little HP amd64 development box--not anything near the kind of machine that you would probably install in a large installation. Asterisk just gets faster and faster. Some of the it isn't good at x stuff comes from experiences with older releases. In a HA and/or high volume scenario I worry about stuff like this that has been in tree since 1.0 or earlier and is in 1.6, channel.c lines 3825~3828: /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ That's not something I want in my high-end, high-capacity, high-availability production system! For smallish installations, this probably isn't a big deal given today's hardware capabilities. Still, it makes me wonder what other gremlins are out there that might bite me in a big-time install. At least with OSS I can see stuff like this. I shudder to think what psycho spaghetti code is running on Cisco, Avaya, Nortel, NEC, Shoretel, etc. -MC If you are lucky enough to have a situation where you can re-invite media and keep it off of the asterisk box, it can handle huge loads. Terry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi wrote: Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Sam Houston University migrated from a Cisco CallManager and Nortel setup to Asterisk a couple years back. I do not know any of the specific details, but maybe you can track down someone involved in the project. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Dan Austin wrote: Yehavi wrote: Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Sam Houston University migrated from a Cisco CallManager and Nortel setup to Asterisk a couple years back. I do not know any of the specific details, but maybe you can track down someone involved in the project. Dan Remember - You are going from a CARRIER GRADE purpose built piece of hardware with Software built under a rigid CMM with extensive soak-testing to software that has been developed under , shall we say, a somewhat less rigid and stringent methodology. You will be moving from an environment supported by hundreds of highly trained people, some with decades of TELCO experience to one where you support comes from a somewhat less seasoned group of individuals. 10,000 extensions ??? On Asterisk ??? You pays your money, you takes you chances. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
