Re: [asterisk-users] asterisk 1.4 with around 230 SIP connections
BJ Weschke wrote: Jerry Geis wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. I would have some serious reservations throwing this many clients into an app_meetme room which is the foundation layer for the page functionality. Well, it may be ok, especially given that the 230 clients are all marked as listen only. There isn't any mixing going on at all. However, there is almost certainly going to be some lag that you may not be happy with. What happens is that you are spawning 230 threads to make outbound calls and connect them to MeetMe, all at the same time. This process is far from instantaneous. :) I would also be concerned about the effects that this spike in extra processing would have on the quality of any existing calls on the system. But, as with most things, the only way to know for sure is to do some testing. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] asterisk 1.4 with around 230 SIP connections
Is it absolutely necessary to have ALL 230 clients get the message at once? Could a few clients in each area be paged to get the announcement to everyone in that area? If these are all soft clients then maybe setting up a recording and then paging groups and having the recording played to smaller groups at one time? Just throwing a couple suggestions out... Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Wednesday, December 12, 2007 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 1.4 with around 230 SIP connections BJ Weschke wrote: Jerry Geis wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. I would have some serious reservations throwing this many clients into an app_meetme room which is the foundation layer for the page functionality. Well, it may be ok, especially given that the 230 clients are all marked as listen only. There isn't any mixing going on at all. However, there is almost certainly going to be some lag that you may not be happy with. What happens is that you are spawning 230 threads to make outbound calls and connect them to MeetMe, all at the same time. This process is far from instantaneous. :) I would also be concerned about the effects that this spike in extra processing would have on the quality of any existing calls on the system. But, as with most things, the only way to know for sure is to do some testing. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] asterisk 1.4 with around 230 SIP connections
On Dec 12, 2007 9:41 AM, Russell Bryant [EMAIL PROTECTED] wrote: BJ Weschke wrote: Jerry Geis wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. I would have some serious reservations throwing this many clients into an app_meetme room which is the foundation layer for the page functionality. Well, it may be ok, especially given that the 230 clients are all marked as listen only. There isn't any mixing going on at all. However, there is almost certainly going to be some lag that you may not be happy with. What happens is that you are spawning 230 threads to make outbound calls and connect them to MeetMe, all at the same time. This process is far from instantaneous. :) I would also be concerned about the effects that this spike in extra processing would have on the quality of any existing calls on the system. But, as with most things, the only way to know for sure is to do some testing. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. Russell, What are your thoughts on SIP/RTP multicast, if any? It's been discussed before. Seems like a great solution for paging (f the phones support it). Anyone interested in a bounty? -- Kristian Kielhofner ___ --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] asterisk 1.4 with around 230 SIP connections
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. Would this work? Thanks, Jerry ___ --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] asterisk 1.4 with around 230 SIP connections
Jerry Geis wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. Would this work? Thanks, Jerry I would have some serious reservations throwing this many clients into an app_meetme room which is the foundation layer for the page functionality. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] asterisk 1.4 with around 230 SIP connections
On Dec 10, 2007 1:17 PM, Jerry Geis [EMAIL PROTECTED] wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. Would this work? Thanks, Jerry I would also be really concerned about the ability for the NIC to serve up all of those RTP streams... 50pps x 230 = 11,500pps It would be nice to have some support for RTP multicast or something. Obviously this would require changes in Asterisk AND support in each phone, but it would be really cool. I think I've seen some Linksys/Sipura devices support it. -- Kristian Kielhofner ___ --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] asterisk 1.4 with around 230 SIP connections
speaking of multi-casting voice. since it isn't likely to get the ip phones changed, could an app_multicast do the job? has anyone thought of doing that? daveC Kristian Kielhofner wrote: On Dec 10, 2007 1:17 PM, Jerry Geis [EMAIL PROTECTED] wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. Would this work? Thanks, Jerry I would also be really concerned about the ability for the NIC to serve up all of those RTP streams... 50pps x 230 = 11,500pps It would be nice to have some support for RTP multicast or something. Obviously this would require changes in Asterisk AND support in each phone, but it would be really cool. I think I've seen some Linksys/Sipura devices support it. -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] asterisk 1.4 with around 230 SIP connections
On Dec 11, 2007 12:19 AM, dave cantera [EMAIL PROTECTED] wrote: speaking of multi-casting voice. since it isn't likely to get the ip phones changed, could an app_multicast do the job? has anyone thought of doing that? daveC Dave, That's just it - I think at least Snom and Linksys/Sipura phones support RTP multicast. Whether it's an app/res/channel/etc all we need is RTP multicast support in Asterisk... -- Kristian Kielhofner ___ --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
