Re: [asterisk-users] Call load balancing
On 9 Mar 2007, at 17:51, Octavio Ruiz (Ta^3) wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Another approach: what about load-balance (in terms of redundancy and scalability) the AGI app's and just the AGIs with FastAGI? So your IVR application can be separated from your * boxes and they (the * boxes) dont have to ve overloaded with your AGI apps. Your head system receive the two PRIs and in dial-plan logic you can (maybe using RANDOM() or something more deterministic like a counter) Assuming the head box takes all the calls you could just use setgroup and getgroupcount on the pri box and use them to count the calls. Using groups has the advantage of dealing with hangup right. The only tricky bit would be implementing min(group) in the dialplan. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Call load balancing
What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Call load balancing
telco servers are Supermicro 2.8GHz P4, 1GB, Digium te410p with 2 PRI plugged in. application server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB, Digium te410p (timing only, all calls over IAX) database server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB No failures in over 2 years. On Fri, 9 Mar 2007, David Ruggles wrote: What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Call load balancing
That's cool, but I doubt my systems could handle that same load ;) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Friday, March 09, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Call load balancing telco servers are Supermicro 2.8GHz P4, 1GB, Digium te410p with 2 PRI plugged in. application server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB, Digium te410p (timing only, all calls over IAX) database server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB No failures in over 2 years. On Fri, 9 Mar 2007, David Ruggles wrote: What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Call load balancing
Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Can you give me a link to more information about how to use the management interface? I've been having a hard time trying to track down more advanced documentation and reference material. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --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] Call load balancing
Never mind I found it shortly after sending this :S Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, March 09, 2007 10:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Call load balancing Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Can you give me a link to more information about how to use the management interface? I've been having a hard time trying to track down more advanced documentation and reference material. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --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] Call load balancing
I've got a system I'm putting together to handle IVR calls with *
I have one head system that terminates two PRIs. It routes the calls from
the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
The * boxes run AGI scripts to process the IVR calls. Can I load balance the
routing if I have five calls each of the IVR * boxes gets two call and the
next call would go to the system that currently has the lowest number of
calls?
Another approach: what about load-balance (in terms of redundancy and
scalability) the AGI app's and just the AGIs with FastAGI? So your
IVR application can be separated from your * boxes and they (the * boxes)
dont have to ve overloaded with your AGI apps.
Your head system receive the two PRIs and in dial-plan logic you can (maybe
using RANDOM() or something more deterministic like a counter)
[just an example]:
exten s,1,Answer
exten s,n,Random(50:next)
exten s,n,AGI(agi://asterisk1/${VAR1}|${VAR2})
exten s,n,Hangup
exten s,n,AGI(agi://asterisk2/${VAR1}|${VAR2})
exten s,n,Hangup
--
Honi soit la vache qui rit.
___
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Re: [asterisk-users] Call load balancing
On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] call load balancing
On 10 Aug 2005, at 16:48, Michiel van Baak wrote:On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote: 1) your provider is voluntarily screwing up VoIP traffic2) some idiot purposingly fills up your pipe with UDP traffic If they fill the pipe with TCP traffic, UDP will be dead aswell. Protocols don't matter, bandwidth does.Actually they do. A smart router/firewall can manage inbound TCP traffic by delaying or dropping outbound acks. This will cause anycorrect TCP implementation to back off.Clearly this isn't perfect, it won't help you if you are being DOS'dbut it will throttle inbound http/smtp.Tim.http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
On 09:15, Thu 11 Aug 05, tim panton wrote: On 10 Aug 2005, at 16:48, Michiel van Baak wrote: On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote: 1) your provider is voluntarily screwing up VoIP traffic 2) some idiot purposingly fills up your pipe with UDP traffic If they fill the pipe with TCP traffic, UDP will be dead as well. Protocols don't matter, bandwidth does. Actually they do. A smart router/firewall can manage inbound TCP traffic by delaying or dropping outbound acks. This will cause any correct TCP implementation to back off. Clearly this isn't perfect, it won't help you if you are being DOS'd but it will throttle inbound http/smtp. Tim. The correct TCP implementation is the key here. If everybody on this world used such implementations a lot of problems would be solved. I seen enough clients relying on timeouts instead of acks etc. I have to admit it will help some, but it will never beat a good QoS agreement with your upstream provider. DOS-attacks are something totally different. It will blow you offline till you contacted your upstream provider and the activated some logic on their switches. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
Jean-Michel Hiver wrote: Dave Redmore wrote: Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm wondering what folks are getting with real world usage... I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Snip ... Jean-Michel, hi; Is that using SIP or IAX2 ? I'd assumed you'd be able to get more than that throughput out of an IAX2 trunk because of the sharing of RTP overhead ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Snip ... Jean-Michel, hi; Is that using SIP or IAX2 ? I'd assumed you'd be able to get more than that throughput out of an IAX2 trunk because of the sharing of RTP overhead ? Using SIP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
Joseph wrote: On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote: I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Of course if you're using g.711 it's a different kettle of fish since it takes 80kbps (g.729 only uses about 24). According to Wiki: G729 is 8Kbps G711 is 64Kbps http://www.voip-info.org/tiki-index.php?page=Codecs Those are theoretical, and don't take into account RTP IP + UDP packet overhead. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote: 1) your provider is voluntarily screwing up VoIP traffic 2) some idiot purposingly fills up your pipe with UDP traffic If they fill the pipe with TCP traffic, UDP will be dead as well. Protocols don't matter, bandwidth does. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call load balancing
Joseph [EMAIL PROTECTED] wrote: On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote: I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Of course if you're using g.711 it's a different kettle of fish since it takes 80kbps (g.729 only uses about 24). According to Wiki: G729 is 8Kbps G711 is 64Kbps http://www.voip-info.org/tiki-index.php?page=Codecs That's the payload. You need to add the IP overhead to those numbers, which will bring the total to what Jean said, above. Trunking your calls over an IAX link will help reduce the total IP overhead. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
Dave Redmore wrote: Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm wondering what folks are getting with real world usage... I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Of course if you're using g.711 it's a different kettle of fish since it takes 80kbps (g.729 only uses about 24). Secondly, assuming I want to replace 8 - 10 POTS lines coming in with someone like NuFone or TELIAX... It would seem to be well worth spending an extra $40/mnth for an extra DSL connection for redundancy (probably with two different ISPs) - so, is there any way to get asterisk to route calls based on the quality of a connection at any given time? If I had 2 DSL connections and Asterisk is registered to 2-3 different service providers - can asterisk do some sort of load balancing of calls? I think you will have less headaches and get better results by dedicating 1 DSL connection to VoIP calls and having the 2nd for all your internet traffic. An ever better way is get some kind of SLA with guaranteed uptime and bandwith, a symetrical link, and do some traffic shaping to ensure that VoIP has priority. Part of the point of VoIP is to save money by collapsing voice and data networks onto one (presumably robust) network, so having 2 shabby separate DSL connections kinds of defeats the purpose. This being said, where symetrical, guaranteed links aren't available or affordable - I'm in this situation :( - ADSL remains a good choice. I have 6 links that are used every day and it works mostly fine. Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call load balancing
--- An ever better way is get some kind of SLA with guaranteed uptime and bandwith, a symetrical link, and do some traffic shaping to ensure that VoIP has priority. Part of the point of VoIP is to save money by collapsing voice and data networks onto one (presumably robust) network, so having 2 shabby separate DSL connections kinds of defeats the purpose. -- How do you traffic shape incoming packets though Without your ISP to provide QoS for downstream voice traffic, quality can still be an issue -Darren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
I am doing traffic shaping with a open source linux firewall http://www.ipcop.org/ and since i have traffic shaping configured my 3 VoIP lines work great. I am not using Asterix yet but I will go to as soon as I have the time to work myself into it. If anybody can tell me where the best information is to get a start on it, I would greatly appreciate it. alex Darren Wright wrote: --- An ever better way is get some kind of SLA with guaranteed uptime and bandwith, a symetrical link, and do some traffic shaping to ensure that VoIP has priority. Part of the point of VoIP is to save money by collapsing voice and data networks onto one (presumably robust) network, so having 2 shabby separate DSL connections kinds of defeats the purpose. -- How do you traffic shape incoming packets though Without your ISP to provide QoS for downstream voice traffic, quality can still be an issue -Darren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
Darren Wright wrote: --- An ever better way is get some kind of SLA with guaranteed uptime and bandwith, a symetrical link, and do some traffic shaping to ensure that VoIP has priority. Part of the point of VoIP is to save money by collapsing voice and data networks onto one (presumably robust) network, so having 2 shabby separate DSL connections kinds of defeats the purpose. -- How do you traffic shape incoming packets though Without your ISP to provide QoS for downstream voice traffic, quality can still be an issue... Well, freebsd's dummynet sort of does it. There is also a product called netequalizer (based on BSD) which seems to do this as well. Second, usually the more bandwith you have, the less you need to shape traffic. Third, usually, doing outgoing shaping and preventing your DSL modem from buiding queues with your upstream traffic improves VoIP tremendously, regardless of inbound traffic shaping. And then, it would be wise to limit TCP traffic to 60-70% of available bandwith. Since UDP / RTP is usually prioritary over TCP, it means you don't really have to worry about inbound shaping, unless: 1) your provider is voluntarily screwing up VoIP traffic 2) some idiot purposingly fills up your pipe with UDP traffic Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call load balancing
On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote: I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Of course if you're using g.711 it's a different kettle of fish since it takes 80kbps (g.729 only uses about 24). According to Wiki: G729 is 8Kbps G711 is 64Kbps http://www.voip-info.org/tiki-index.php?page=Codecs -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
