[asterisk-users] Bandwidth management and ADSL router
Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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] Bandwidth management and ADSL router
A lot of the ADSL CPE (customer premise equipment) deployed has basic QoS capabilities in a pre-set kind of way, but if you want to do your own DiffServ tagging the standard practice is to do Layer 2 Ethernet bridging to a more intelligent box behind the ADSL CPE. bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Bandwidth management and ADSL router
m0n0wall and pfsense both do traffic shaping, which forcibly allocates bandwidth for your VoIP traffic. Michael On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] Bandwidth management and ADSL router
On Tue, 26 May 2009, bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Draytek 2800 series routers have adequate traffic management on outgoing traffic to do a reasonable job. (There is a very little you can do to shape incoming traffic) However you need to make sure that the actual Internet connection isn't where the bottleneck is. Try making calls when you can guarantee that no other traffic is flowing into/out of each end. 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] Bandwidth management and ADSL router
As does ZeroShell (www.zeroshell.net/eng). Bruce Komito WPTI Telecom (775) 236-5815 On Tue, 26 May 2009, Michael Graves wrote: m0n0wall and pfsense both do traffic shaping, which forcibly allocates bandwidth for your VoIP traffic. Michael On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] Bandwidth management and ADSL router
Thanks for all. But what all gave me was a software need to be installed on PC, but I am looking for a router (ADSL router) that can does this, because usually the ADSL router is the default gateway where all the traffic goes out and in. Any ADSL router device can do this? About Draytek, as I understand that control can be done only at upload traffic and not download traffic, while 90% of the problem are coming from download traffic, so this is not the needed. Any advise in that direction? Regards Bilal ___ -- 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] Bandwidth management and ADSL router
A lot of ISP adsl modems aren't capable. But most should be. Just log in to it give it a private IP for your lan(192.X.X.X or whatever your using) then have all your computers use that local IP as their gateway address. If you have an ADSL modem which doesn't then simple get a router (hell a Linksys/Dlink $50 cheapy from wallmart would work) and have the ADSL plug into the router and all the stations use the router for their gateway. If you have a spare server or virtual server space you can use Vyatta (Vyatta.com) it is a free open source router/firewall/vpn/few other things. I've never used it in a virtual environment, but I see no reason why it wouldn't work that way. Also note that it requires almost nothing to run so you can put it on an old 1Ghz machine and It would still operate just fine. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein Theory is when you know something, but it doesn't work. Practice is when something works, but you don't know why. Programmers combine theory and practice: Nothing works and they don't know why.-Anonymous Developer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, May 26, 2009 11:55 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Bandwidth management and ADSL router Thanks for all. But what all gave me was a software need to be installed on PC, but I am looking for a router (ADSL router) that can does this, because usually the ADSL router is the default gateway where all the traffic goes out and in. Any ADSL router device can do this? About Draytek, as I understand that control can be done only at upload traffic and not download traffic, while 90% of the problem are coming from download traffic, so this is not the needed. Any advise in that direction? Regards Bilal ___ -- 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] Bandwidth management and ADSL router
I've had good luck using a sangoma S518 ADSL card in a linux box. the logging capabilities are supurb (cought my provider not providing what they said they were and great for troubleshooting as it logs line speed and dropouts to the second). support is also top notch. once installed it looks to the system like any other interface. Since it looks to the system like any other interface you have the full power of routing, bridging, firewalling, iptables, neumerous queing schemes, etc. everything linux has to offer. It has served me well and is extremely flexable. Eric Fort FortConsulting On Tue, May 26, 2009 at 4:32 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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] Bandwidth management and ADSL Router
Dear Eric; Sangoma has ADSL router? And does that router support bandwidth division capability? Dear jas; About what u mentioned: it is related to linux, do u know a dsl router that does bandwidth divion? Any help? Regards Bilal I've had good luck using a sangoma S518 ADSL card in a linux box. the logging capabilities are supurb (cought my provider not providing what they said they were and great for troubleshooting as it logs line speed and dropouts to the second). support is also top notch. once installed it looks to the system like any other interface. Since it looks to the system like any other interface you have the full power of routing, bridging, firewalling, iptables, neumerous queing schemes, etc. everything linux has to offer. It has served me well and is extremely flexable. Eric Fort FortConsulting On Tue, May 26, 2009 at 4:32 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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] Bandwidth management and ADSL Router
bilal ghayyad wrote: Dear Eric; Sangoma has ADSL router? And does that router support bandwidth division capability? Internal ADSL cards; not external router appliances. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] bandwidth required for Asterisk running on T1
Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? Thanks in advance for your input. Regards, Mark ___ -- 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] bandwidth required for Asterisk running on T1
mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- 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] bandwidth required for Asterisk running on T1
Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- 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 ___ -- 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] bandwidth required for Asterisk running on T1
Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? Thanks in advance for your input. Regards, Mark Check out http://www.asteriskguru.com/tools/bandwidth_calculator.php it should help you figure out how much bandwidth you will need. Ryan ___ -- 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] bandwidth required for Asterisk running on T1
That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- 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 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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] bandwidth required for Asterisk running on T1
Hi Andrew, Yes, it is actually a E1. Your suggestion will be greatly appreciated. Thanks, Mark On Fri, Apr 11, 2008 at 7:50 AM, Andrew Latham [EMAIL PROTECTED] wrote: That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- 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 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.com http://www.tuxtone.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 ___ -- 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] bandwidth required for Asterisk running on T1
Using the online calculator mentioned in this thread will help. There is a lot to bandwidth and even more to VoIP network traffic than can be answered with your question. On an E1 that is dedicated to IAX terminating to a provider that does trunking I would say that you could get a large number of concurrent calls through On the other hand if the calls where SIP u.law and going to different network destinations you may only get a few concurrent calls to work. Its like a good bottle of wine, the bottle is just the container On Fri, Apr 11, 2008 at 11:15 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi Andrew, Yes, it is actually a E1. Your suggestion will be greatly appreciated. Thanks, Mark On Fri, Apr 11, 2008 at 7:50 AM, Andrew Latham [EMAIL PROTECTED] wrote: That sounds like an E1 to me. Is that 32 DS0 channels or 24? On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- 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 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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 ___ -- 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.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] bandwidth required for Asterisk running on T1
On Fri, 2008-04-11 at 01:18 -0700, mark morreny wrote: The T1 is 32 x 64Kbps channels ; Codec is GSM. That's incorrect... a T1 is 24 channels, and each channel is 64kbps. There are also a few extra bits for framing, which adds up to 1.544 megabits per second in each direction. The audio comes across a T1 as G.711 (not GSM as stated above), and on a T1 it's usually using ulaw companding. An E1 is 32 channels, and each channel is the same 64kbps. This adds up to 2.048 megabits per second. Again, the audio is in G.711 format, but alaw companding is typically used on an E1. -- Jared Smith Community Relations Manager 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] bandwidth required for Asterisk running on T1
Additionally Mark, a Channelized (also called Integrated) T1 offers 24 channels for voice/data, but after bit robbing (for signalling, etc) you only get around 56kbps per channel. ISDN PRI over T1 has 23 b-channels of voice/data and one d-channel for signalling, etc. PRI is preferred and most common. And of course, ISDN PRI over E1 gets 30 channels of voice/data and 2 channels for signalling. Jared Smith wrote: On Fri, 2008-04-11 at 01:18 -0700, mark morreny wrote: The T1 is 32 x 64Kbps channels ; Codec is GSM. That's incorrect... a T1 is 24 channels, and each channel is 64kbps. There are also a few extra bits for framing, which adds up to 1.544 megabits per second in each direction. The audio comes across a T1 as G.711 (not GSM as stated above), and on a T1 it's usually using ulaw companding. An E1 is 32 channels, and each channel is the same 64kbps. This adds up to 2.048 megabits per second. Again, the audio is in G.711 format, but alaw companding is typically used on an E1. -- Jared Smith Community Relations Manager 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
[asterisk-users] Bandwidth and Colocation (was: Re: Is it possible to use spandsp and patton to do fax2mail ?)
Robert Moskowitz wrote: ___ -- 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 9 times? (Ok, I cannot blame you for the 9th one.) Gee. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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] Bandwidth shapping device
I would use a Mikrotik - www.mikrotik.com Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, February 14, 2007 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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] Bandwidth shapping device
Quoting Ronald Wiplinger [EMAIL PROTECTED]: I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) install openbsd on some old hardware with 2 x nics in it. use a bridge configuration with no ips, and use the pf traffic shaping rules to split it up however you want. you don't have to just dedicate chunks of the bandwidth, you can setup limits, but still let them borrow from other non-full peer channels as well. One setup like this at either end will manage the traffic in both directions through the link. openbsd is a little known operating system that focuses on security above all else, and its the perfect tool for routers/firewalls/traffic shapers, etc. you can generate the pf configurations with fwbuilder from linux or windows, using a gui instead of hand editing the file, but I am not sure if the the traffic shaping features are supported there or not, never tried it. bye Ronald Wiplinger ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Bandwidth shapping device
On 22:19, Wed 14 Feb 07, Ronald Wiplinger wrote: I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) I second Jon Pounder's advice. Get an OpenBSD device. You dont need 2 boxes, you can shape on both nics. That way one machine is enough. Here's the official FAQ about queueing in OpenBSD: http://www.openbsd.org/faq/pf/queueing.html -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Bandwidth shapping device
I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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] Bandwidth shapping device
Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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] Bandwidth shapping device
On Wed, 14 Feb 2007, Damon Estep wrote: Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. I have to say, that unless you are quite good at driving Linux or *BSD's firewall/traffic shaping mechanisms, then I'd probably go for a Cisco - especially if this is a full-on corp-rat environment. I would use a Linux box, but then I've been using Linux boxes for a great number of years including setting up some hairy/scarey traffic management. Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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] Bandwidth shapping device
Try Mikrotik RouterOS. They are quite str8 fwd to config and very very versatile. Documentation is also available. That would be a piece of cake to implement. Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 14 Feb 2007, Damon Estep wrote: Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. I have to say, that unless you are quite good at driving Linux or *BSD's firewall/traffic shaping mechanisms, then I'd probably go for a Cisco - especially if this is a full-on corp-rat environment. I would use a Linux box, but then I've been using Linux boxes for a great number of years including setting up some hairy/scarey traffic management. Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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 - What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship.___ --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] Bandwidth shapping device
Hi, What Network Switch you are using? I do traffic/bandwidth shapping on the edge switch where the port the voice installed, you can configure each port to 128Kbps or just plain Ethernet Port. So the link between bldg. will always be 10Mb/s, who ever uses it whether data or voice and enable switch port prioritization. HP ProCurve and Cisco switches do this features. Don't know if others can do it. Regards Angel Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 14 Feb 2007, Damon Estep wrote: Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. I have to say, that unless you are quite good at driving Linux or *BSD's firewall/traffic shaping mechanisms, then I'd probably go for a Cisco - especially if this is a full-on corp-rat environment. I would use a Linux box, but then I've been using Linux boxes for a great number of years including setting up some hairy/scarey traffic management. Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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 - Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains.___ --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] Bandwidth shapping device
Hello I use trafic shapper, is very good. Regards On 2/14/07, Angel Heart [EMAIL PROTECTED] wrote: Hi, What Network Switch you are using? I do traffic/bandwidth shapping on the edge switch where the port the voice installed, you can configure each port to 128Kbps or just plain Ethernet Port. So the link between bldg. will always be 10Mb/s, who ever uses it whether data or voice and enable switch port prioritization. HP ProCurve and Cisco switches do this features. Don't know if others can do it. Regards Angel *Gordon Henderson [EMAIL PROTECTED]* wrote: On Wed, 14 Feb 2007, Damon Estep wrote: Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. I have to say, that unless you are quite good at driving Linux or *BSD's firewall/traffic shaping mechanisms, then I'd probably go for a Cisco - especially if this is a full-on corp-rat environment. I would use a Linux box, but then I've been using Linux boxes for a great number of years including setting up some hairy/scarey traffic management. Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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 -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --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 -- Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel siteshttp://farechase.yahoo.com/promo-generic-14795097;_ylc=X3oDMTFtNW45amVpBF9TAzk3NDA3NTg5BF9zAzI3MTk0ODEEcG9zAzEEc2VjA21haWx0YWdsaW5lBHNsawNxMS0wNw--%0Ato find flight and hotel bargains. ___ --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] Bandwidth requirements for 1, 000, 000 minutes a month
I second that Luki. We at www.cyberdyne-ip.com (yes shameless plug) only use ulaw for termination. Of course we have to offer g729, GSM, etc. to our customers... but for best quality, we transcode to ulaw if we send the call to another carrier for termination. 729 may use less bandwidth and in turn cost less, but what is more important... cost or call quality? My 2 cents, bp On 12/15/06, Luki [EMAIL PROTECTED] wrote: But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P I do. Exclusively. I personally don't like the g729 compression (audio quality and license issues) any my customers definitely notice the difference right away and wonder why the quality degraded. I guess I spoiled them with ulaw. So no g729 here. g726-32 on the other hand was acceptable, although the difference is still noticeable. --Luki ___ --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] Bandwidth requirements for 1, 000, 000 minutes a month
This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The busiest hour will account for 10% of the daily total. This is how I'm figuring it... Casual observation shows that SIP setup and teardown takes about 26 UDP packets. Assuming the packets are full (512 bytes) this adds up to about 13 kilo-bytes for each call. I've heard that ulaw (including overhead) is supposed to take about 80 kilo-bits/sec. Assuming each call takes 10 minutes, each call will take 13 kilo-bytes + (80 kilo-bits * 60 * 10) or 48.13 mega-bits. Assuming (to make the math easier) 10 bits = 1 byte, each call will take 4.813 mega-bytes. So, 100,000 calls of 10 minutes (1 million minutes) would consume 481,300 mega-bytes per month or 3,333 calls consuming 16,043 mega-bytes per day. Assuming the busiest hour accounts for about 10% of the daily total, that hour would consist of 333 calls consuming 1,604 mega-bytes. So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Would anybody venture an estimate of what the peak bandwidth would be if we changed to IAX? With trunking? 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] Bandwidth requirements for 1, 000, 000 minutes a month
But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P gsm, ilbc, g729 etc are a lot better choice. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Steve Edwards wrote: This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The busiest hour will account for 10% of the daily total. This is how I'm figuring it... Casual observation shows that SIP setup and teardown takes about 26 UDP packets. Assuming the packets are full (512 bytes) this adds up to about 13 kilo-bytes for each call. I've heard that ulaw (including overhead) is supposed to take about 80 kilo-bits/sec. Assuming each call takes 10 minutes, each call will take 13 kilo-bytes + (80 kilo-bits * 60 * 10) or 48.13 mega-bits. Assuming (to make the math easier) 10 bits = 1 byte, each call will take 4.813 mega-bytes. So, 100,000 calls of 10 minutes (1 million minutes) would consume 481,300 mega-bytes per month or 3,333 calls consuming 16,043 mega-bytes per day. Assuming the busiest hour accounts for about 10% of the daily total, that hour would consist of 333 calls consuming 1,604 mega-bytes. So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Would anybody venture an estimate of what the peak bandwidth would be if we changed to IAX? With trunking? 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 Inbound (clean). Database: 0659-0, 12/15/2006 - 12/15/2006 9:47:48 PM ___ --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] Bandwidth requirements for 1, 000, 000 minutes a month
So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Sounds about right. Or the other way around (if you need to know the peak bandwidth usage): For audio: 1,000,000 minutes/month = 33,000 minutes/day 10% daily usage in 1 hour = 3,300 minutes used 3,300 minutes used in 60 minutes = 55 concurrent calls 80 kbps / 1 call direction * 55 calls = 4.4 Mbps per direction Assuming full duplex audio, you need 4.4 Mbps in + 4.4 Mbps out per call leg. If you route the call so each packet comes in and goes out the network (2 call legs), then double the bandwidth. I guess adding 0.1 Mbps for call setup and tear down is safe. --Luki ___ --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] Bandwidth requirements for 1, 000, 000 minutes a month
But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P I do. Exclusively. I personally don't like the g729 compression (audio quality and license issues) any my customers definitely notice the difference right away and wonder why the quality degraded. I guess I spoiled them with ulaw. So no g729 here. g726-32 on the other hand was acceptable, although the difference is still noticeable. --Luki ___ --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] Bandwidth requirements
Let me paste my old reply to this: Let's do some calculations on that: g729a 20ms results in 20 bytes RTP payload in each packet, in order to traverse the OSI model there's some headers that need to be added, as RTP gets an RTP header, the RTP packet gets an UDP header and the UDP datagram gets an IP header: So add a 12 byte RTP header, 8 byte UDP header and a 20 Byte IP header This results in: 20 byte RTP payload + 12 byte RTP header + 8 byte UDP header + 20 byte IP header=60 byte on the ip layer. Thats a 40 byte overhead (so 2/3 of the packet is just headers :) and were still only on the IP layer now) So to transmit just 1 RTP packet you are actualy transmitting 60 bytes on the IP layer, so in order to get the real used bandwidth we need to knowhow many packets we are sending and on which medium (DSL/ethernet/slip/smokesignals): 20 ms results in 50 packets/s so: 50 packet/s *60 bytes/packet=3000 bytes/s that's 300*8=24000 bit/s total bandwidth on the IP layer so the overhead is 24000-8000=16000 bit/s. The fun starts if you are going to send this over DSL, let's continue the calculation: 50 packets of 60 byte IP, add the 2 byte PPPoA header for DSL= 62 bytes per packet. However, DSL operates with 53 bytes ATM cells, in which you can fit 48 bytes payload (and a 5 byte header) so in order to transmit the 62 bytes of data you need: 62/48=2 ATM cells. Why 2 cells you say? Because ATM can't utilize the unused part of cells, so to transmit 62 bytes you use the same amount of bandwith (on dsl) as you would use to transmit 96 (48*2) bytes. So 1 RTP packet uses 96 bytes on the DSL line, as you already know we have 50 packets/s so that's 50 packets/s*2 cells=100 Cells/s 100 cells/s * 53 byte = 5300 bytes/s on the DSL line thats 42400 bits/s to transmit a 8 kbit/s stream :) So in order to use 5 simultaneous calls you would need a 1:1 DSL line of at least 5*42400=212000 bps so a 256/256 DSL would do, however if you need 20 calls that would be 20*42400=848000 bps so that would be a 1M/1M line (and some bandwidth to spare) Erik Versaevel hugolivude wrote: Hi, Age old question it seems but I haven't been able to get a handle on it yet. Let's assume I'm using a g729 codec. If I wanted to handle 20 simultaneous calls, how much bandwidth would I need? Is there a general formula for this? I tried this caluclator: http://www.voip-calculator.com/calculator/eipb/ I wasn't sure what Packet Duration to select so I took the default 20ms (2 samples) - whatever that means. I plugged in 5 for the BHT (20 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs) and the default 0.01 for the Blocking. It worked out to 264 kbps. Does this sound reasonable? If so great! A business DSL could support this. Comments welcome! Cheers, H ___ --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 Erik Versaevel ___ --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] Bandwidth requirements
On Wed, Oct 04, 2006 at 07:51:14PM -0400, hugolivude wrote: Age old question it seems but I haven't been able to get a handle on it yet. Let's assume I'm using a g729 codec. If I wanted to handle 20 simultaneous calls, how much bandwidth would I need? Is there a general formula for this? I tried this caluclator: [1]http://www.voip-calculator.com/calculator/eipb/ I wasn't sure what Packet Duration to select so I took the default 20ms (2 samples) - whatever that means. A 20ms packet duration means that 20ms of audio is stuffed into one IP packet. Since each packet carries 1/50th of a second of audio, that means you're generating 50 packets per second for each channel. With g729 your audio is 8000 bits per second. The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes (RTP) = 40 bytes or 320 bits. So your bandwidth requirement per channel is: - 8000 bits per second for payload - 320x50 = 16000 bits per second for overhead making a total of 24000 bits per second. 20 simultaneous calls is therefore 480,000 bits per second. That is a bit of an underestimate though, because it doesn't include any layer 2 framing overhead (i.e. for encapsulating the IP frames in the underlying medium). For example, if it were HDLC serial on a leased line, that would be just 2 bytes per frame for flags, maybe a couple of bytes for CRC, plus occasional bit-stuffing. However on ADSL, you have to add the 15% ATM cell tax. And you would be wise to add 20% headroom (i.e. so your line is not more than 80% full) As you can see, the packetisation overhead is twice as large as the useful data you're transporting. You can reduce this by increasing the packet duration, but that increases the latency of your audio (and ADSL links already add 20-30ms of latency themselves). Too much latency is objectionable to users. I have read that if you use IAX2 trunking it's able to combine audio from multiple streams into a single packet, thus sharing the overhead between them, but I have no experience of this myself. HTH, Brian. ___ --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] Bandwidth requirements
Brian Candler wrote: However on ADSL, you have to add the 15% ATM cell tax. And you would be wise to add 20% headroom (i.e. so your line is not more than 80% full) ATM cell tax is actually 10% as there's 5 header bytes for each 53 bytes cell, ___ --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] Bandwidth requirements
On Thu, Oct 05, 2006 at 09:44:56AM +0100, Brian Candler wrote: A 20ms packet duration means that 20ms of audio is stuffed into one IP packet. Since each packet carries 1/50th of a second of audio, that means you're generating 50 packets per second for each channel. With g729 your audio is 8000 bits per second. The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes (RTP) = 40 bytes or 320 bits. So your bandwidth requirement per channel is: - 8000 bits per second for payload - 320x50 = 16000 bits per second for overhead making a total of 24000 bits per second. 20 simultaneous calls is therefore 480,000 bits per second. A reminder: much equipment, particularly low end/consumer equipment, chokes *much* faster on high PPS than it does on high BPS. While short packets are good for latency, they do impose stricter engineering evaluation requirements on the other links in your chains. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] Bandwidth requirements
For bandwidth requeriments don't forget Layer 2 overhead. I.e Frame-relay overhead is lower than Ethernet overhead. Rgds. On 10/5/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Oct 05, 2006 at 09:44:56AM +0100, Brian Candler wrote: A 20ms packet duration means that 20ms of audio is stuffed into one IP packet. Since each packet carries 1/50th of a second of audio, that means you're generating 50 packets per second for each channel. With g729 your audio is 8000 bits per second. The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes (RTP) = 40 bytes or 320 bits. So your bandwidth requirement per channel is: - 8000 bits per second for payload - 320x50 = 16000 bits per second for overhead making a total of 24000 bits per second. 20 simultaneous calls is therefore 480,000 bits per second. A reminder: much equipment, particularly low end/consumer equipment, chokes *much* faster on high PPS than it does on high BPS. While short packets are good for latency, they do impose stricter engineering evaluation requirements on the other links in your chains. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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 -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --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] Bandwidth requirements
On Thu, Oct 05, 2006 at 10:53:14AM +0200, raphael Jacquot wrote: Brian Candler wrote: However on ADSL, you have to add the 15% ATM cell tax. And you would be wise to add 20% headroom (i.e. so your line is not more than 80% full) ATM cell tax is actually 10% as there's 5 header bytes for each 53 bytes cell, Only if all cells are filled. On average there will be half a cell empty at the end of each packet. A common case is 1500-byte packets; these will take 32 cells, or 1696 bytes total, giving a tax of 13%. Mix some smaller packets in with that and you get a higher tax. So I tend to work on 15% as a rule of thumb. However it's worse for VoIP as has been pointed out. e.g. if you are sending 60 byte packets (20 IP, 8 UDP, 12 RTP, 20 G729) then they will take two 53-byte cells, so you pay 46 extra bytes to carry 60 bytes of IP; the tax is then 77% (You will also have encapsulation, e.g. PPPoA, but that probably fits in the wasted space without needing another cell) Regards, Brian. ___ --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] Bandwidth requirements
So much detail! Thanks very much guys, I'm sure that all this excellent info will be valuable to others as well. Gratefully yours, HOn 10/5/06, Brian Candler [EMAIL PROTECTED] wrote: On Thu, Oct 05, 2006 at 10:53:14AM +0200, raphael Jacquot wrote: Brian Candler wrote: However on ADSL, you have to add the 15% ATM cell tax. And you would be wise to add 20% headroom ( i.e. so your line is not more than 80% full) ATM cell tax is actually 10% as there's 5 header bytes for each 53 bytes cell,Only if all cells are filled. On average there will be half a cell empty at the end of each packet.A common case is 1500-byte packets; these will take 32 cells, or 1696 bytestotal, giving a tax of 13%. Mix some smaller packets in with that and youget a higher tax. So I tend to work on 15% as a rule of thumb. However it's worse for VoIP as has been pointed out. e.g. if you are sending60 byte packets (20 IP, 8 UDP, 12 RTP, 20 G729) then they will take two53-byte cells, so you pay 46 extra bytes to carry 60 bytes of IP; the tax is then 77%(You will also have encapsulation, e.g. PPPoA, but that probably fits in thewasted space without needing another cell)Regards,Brian.___ --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
[asterisk-users] Bandwidth requirements
Hi,Age old question it seems but I haven't been able to get a handle on it yet.Let's assume I'm using a g729 codec.If I wanted to handle 20 simultaneous calls, how much bandwidth would I need?Is there a general formula for this?I tried this caluclator: http://www.voip-calculator.com/calculator/eipb/I wasn't sure what Packet Duration to select so I took the default 20ms (2 samples) - whatever that means. I plugged in 5 for the BHT (20 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs) and the default 0.01 for the Blocking. It worked out to 264 kbps. Does this sound reasonable? If so great! A business DSL could support this. Comments welcome! Cheers, H ___ --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] Bandwidth requirements
Hi, Age old question it seems but I haven't been able to get a handle on it yet.Let's assume I'm using a g729 codec.If I wanted to handle 20 simultaneous calls, how much bandwidth would I need?Is there a general formula for this? I tried this caluclator: http://www.voip-calculator.com/calculator/eipb/ I wasn't sure what Packet Duration to select so I took the default 20ms (2 samples) - whatever that means. I plugged in 5 for the BHT (20 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs) and the default 0.01 for the Blocking. It worked out to 264 kbps. Does this sound reasonable? If so great! A business DSL could support this. Comments welcome! Cheers, H ___ --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] Bandwidth requirements
I wasn't sure what Packet Duration to select so I took the default 20ms (2 samples) - whatever that means. I plugged in 5 for the BHT (20 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs) and the default 0.01 for the Blocking. It worked out to 264 kbps. Does this sound reasonable? If so great! A business DSL could support this. Comments welcome! Sounds about right to me. 8kbps + overhead. Transcoding 20 simultaneous channels requires a relatively hefty machine I think ___ --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] Bandwidth usage
Hello, I thought I'd ask this, just in case I'm wrong. We're trying to set up 'remote' users via asterisk. Basically all there is to this is asterisk forwarding a DID to a cell phone. My question is this: Is there any possible way for our local asterisk box to setup the connection and the drop out so that our bandwidth isn't being used for the call? I really don't think this is at all possible but I thought I would double check. Perhaps with the cooperation of your DID provider? ___ --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] Bandwidth usage
On Tue, Oct 03, 2006 at 10:33:38AM -0400, Steve Glaus wrote: I thought I'd ask this, just in case I'm wrong. We're trying to set up 'remote' users via asterisk. Basically all there is to this is asterisk forwarding a DID to a cell phone. My question is this: Is there any possible way for our local asterisk box to setup the connection and the drop out so that our bandwidth isn't being used for the call? I really don't think this is at all possible but I thought I would double check. Perhaps with the cooperation of your DID provider? You can do this if your trunks are PRI, and your carrier supports TBCT (two B-channel transfer)... or whatever they're calling it these days. I saw this feature documented in the NT DMS-100 feature planning guide as much as 10 years ago; I assume a) it's still available and b) carriers have finally started to roll it out. :-) But at least you know what to ask for. Remember you won't get call-detail on those calls. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] Bandwidth usage
Jay R. Ashworth wrote: On Tue, Oct 03, 2006 at 10:33:38AM -0400, Steve Glaus wrote: I thought I'd ask this, just in case I'm wrong. We're trying to set up 'remote' users via asterisk. Basically all there is to this is asterisk forwarding a DID to a cell phone. My question is this: Is there any possible way for our local asterisk box to setup the connection and the drop out so that our bandwidth isn't being used for the call? I really don't think this is at all possible but I thought I would double check. Perhaps with the cooperation of your DID provider? You can do this if your trunks are PRI, and your carrier supports TBCT (two B-channel transfer)... or whatever they're calling it these days. I saw this feature documented in the NT DMS-100 feature planning guide as much as 10 years ago; I assume a) it's still available and b) carriers have finally started to roll it out. :-) But at least you know what to ask for. Remember you won't get call-detail on those calls. Last time I checked... the PRI providers that my customers are using *do* support TBCT. However, the last time I checked, Asterisk did not. Maybe things for Asterisk have changed, though. Lee. ___ --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] Bandwidth via my Asterisk PBX
Am I correct in assuming that all calls from each organization would route through our Asterisk server be passed off to the service provider That depends on your setup, on the provider and on the organization. If all support ReInvites and have them enabled, then it will work and the RTP traffic will flow between the organization and the provider. But each organization needs to be reachable via a public IP (i.e. not NAT) and the provider you use must support it too. I believe most(?) do, but you should check. --Luki ___ --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] Bandwidth via my Asterisk PBX
Am new to Asterisk - have it up and running connected to a couple service providers (telasip teliax). Nice! Am running our Asterisk PBX on a static DSL connection (~600 kbps up/~4mbps down), and would like to extend VoIP service to 10 non-profits we're working with. Am I correct in assuming that all calls from each organization would route through our Asterisk server be passed off to the service provider ( i.e. TelaSip, Teliax) thus keeping our bandwidth requirements to a minimum? Or - are ALL calls that route through our Asterisk PBX consuming our bandwidth through the duration of the call? Thanks, Dakota ___ --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] Bandwidth Management
I think this belongs to the development mail-list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, April 12, 2006 12:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bandwidth Management Andy Tan a écrit : Hi Alex, thanks for the suggestion. Did some checks, and thought that I could set a global variable to track the utilized bandwidth. Wish that there are plans for support to include variables like SIP_CODEC in other protocols. Actually this sounds like a really nice idea. It would be cool to have a way to start using less intensive bandwith codecs for new calls when bandwith reaches a certain threshold. For example: - 0-40% bandwith: g711 - 40-60% bandwith: g729 - 60%-80% bandwith: g723 - 80%-100% bandwith: drop new calls, or maybe use lpc10 It wouldn't help in SOHO usage but when using Asterisk as a call termination gateway, it would help making the most out of available bandwith. g711 is certainly better than g729 when you have the bandwith, and i'm pretty sure that even lpc10 sounds better when on non-saturated bandwith compared with g729 with some packet loss... How would you go about implementing this? Cheers, Jean-Michel. -- 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 ___ --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] Bandwidth Management
Brought over from -users, Please reply to the -dev list.
I agree, lets move the discusstion over to that list as it has to be discussed
there. After we reach an accord on how it should be done we will open up a
issue on Mantis.
I see this as being two distinctive parts that would need to be tied together:
First: We need to make the selection of CODECS technology agnostic, There
currently exist a facility for CODEC selection (SIP_CODEC) in the sip channel
but not in others.
Second: Discuss is this sould be an outside application that is called from
within Asterisk or if it should become a function
Set(CODEC=${OPTIMALCODEC(quality)})
available options could be:
quality
bandwidth
license
Any comments.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Wednesday, April 12, 2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Bandwidth Management
I think this belongs to the development mail-list.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Wednesday, April 12, 2006 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bandwidth Management
Andy Tan a écrit :
Hi Alex,
thanks for the suggestion.
Did some checks, and thought that I could set a global variable to
track the utilized bandwidth.
Wish that there are plans for support to include variables like
SIP_CODEC in other protocols.
Actually this sounds like a really nice idea. It would be
cool to have a way to start using less intensive bandwith
codecs for new calls when bandwith reaches a certain threshold.
For example:
- 0-40% bandwith: g711
- 40-60% bandwith: g729
- 60%-80% bandwith: g723
- 80%-100% bandwith: drop new calls, or maybe use lpc10
It wouldn't help in SOHO usage but when using Asterisk as a
call termination gateway, it would help making the most out
of available bandwith. g711 is certainly better than g729
when you have the bandwith, and i'm pretty sure that even
lpc10 sounds better when on non-saturated bandwith compared
with g729 with some packet loss...
How would you go about implementing this?
Cheers,
Jean-Michel.
--
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
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[Asterisk-Users] Bandwidth Management
Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - Does exactly what it says on the tin ___ --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] Bandwidth Management
Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - mmm... Fastmail... ___ --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] Bandwidth Management
Out of the Box probably not but with an AGI script this is very doable: You can have a script that monitors active calls and the Codecs that are in use. The script will have to do some math to calculate the bandwidth in use and then using the variables in Asterisk, Namely SIP_CODEC. If you are using SIP. There has not been a Variable coded for the other Technologies at this time. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Tan Sent: Tuesday, April 11, 2006 9:00 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Bandwidth Management Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - mmm... Fastmail... ___ --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] Bandwidth Management
On 4/11/06, Andy Tan [EMAIL PROTECTED] wrote: Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan To the best of my knowledge, Asterisk does not have such a feature at the current time. -Rusty ___ --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] Bandwidth Management
Hi Alex, thanks for the suggestion. Did some checks, and thought that I could set a global variable to track the utilized bandwidth. Wish that there are plans for support to include variables like SIP_CODEC in other protocols. Regards On Tue, 11 Apr 2006 12:50:56 -0400, Alexander Lopez [EMAIL PROTECTED] said: Out of the Box probably not but with an AGI script this is very doable: You can have a script that monitors active calls and the Codecs that are in use. The script will have to do some math to calculate the bandwidth in use and then using the variables in Asterisk, Namely SIP_CODEC. If you are using SIP. There has not been a Variable coded for the other Technologies at this time. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Tan Sent: Tuesday, April 11, 2006 9:00 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Bandwidth Management Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - mmm... Fastmail... ___ --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 -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - And now for something completely different ___ --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] Bandwidth Management
Andy Tan a écrit : Hi Alex, thanks for the suggestion. Did some checks, and thought that I could set a global variable to track the utilized bandwidth. Wish that there are plans for support to include variables like SIP_CODEC in other protocols. Actually this sounds like a really nice idea. It would be cool to have a way to start using less intensive bandwith codecs for new calls when bandwith reaches a certain threshold. For example: - 0-40% bandwith: g711 - 40-60% bandwith: g729 - 60%-80% bandwith: g723 - 80%-100% bandwith: drop new calls, or maybe use lpc10 It wouldn't help in SOHO usage but when using Asterisk as a call termination gateway, it would help making the most out of available bandwith. g711 is certainly better than g729 when you have the bandwith, and i'm pretty sure that even lpc10 sounds better when on non-saturated bandwith compared with g729 with some packet loss... How would you go about implementing this? Cheers, Jean-Michel. -- 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] Bandwidth: to seperate or not to seperate
Hi Rich, Thanks for replying to this question - the decision is confusing me a lot :) You said: "Help us understand exactly what this "incoming traffic flooding the bandwidth" is suppose to mean. Are you running something else besides web and voip through this link? If not, then what is "flooding" your bandwidth?" You are right about web page serving not using much incoming bandwidth (good for one-sided QoS management). I was inaccurate by saying "hosting webpages" - we also have all email traffic and host racked servers for customers of ours too (and I have them on a lower QoS in the Netfilter/Wondershaper setup). Actually, typically, the servers are business customers and probably don't use much bandwidth at all but I can't be sure that one of them would not, at any time, upload data from their mega-high-speed office connection - its a bit of an unknown. It's a bit of a bummer that inbound traffic shaping cannot be done - considering that its a data-center setup as opposed to an office/home setup (kind of a so-close-but-so-far type thing). Thinking about it now, before paying money for something we don't need, I should probably try to graph bandwidth usage (which opens a can of worms too - I'm used to MRTG but it would only show a 5 minute average and not instant peaks that would affect VoIP quality - have you ever used any other graphing tools?). I remember when I first started playing with Netfilter TC for QoS I was really surprised to find that very few people seem concerned with QoS routing which is amazing. Thanks again for your opinions! Derek Rich Adamson wrote: Inline... RE: Bandwidth. We have an asterisk server sharing bandwidth with other [web] servers in cabinets that we rent in a large data-center and all is working fine. But I'm concerned that web traffic could affect the VoIP quality (my tests so far haven't showed this [yet!]. Currently I'm running a server with Netfilter (iptables) between all the servers and the Internet with Forward rules and I'm also including a "wondershaper" type QoS ruleset with TC to priortise the outgoing VoIP traffic (I say outgoing because this is really the only thing I can shape on the connection as far as I can see). If your web server is oriented around simply serving up static pages with no one "uploading" data to it, then the majority of the web traffic will be outbound traffic. (eg, user clicks on a link, small amount of inbound traffic to communicate that click, followed by lots of outbound traffic reppreenting the new page(s) to be viewed.) The wondershaper function should prrioritze that mix of traffic just fine. My choice, going forward, is to either buy more bandwidth and magically implement better QoS or the other option is to bring in a separate patch cable, with separate bandwidth, and a different IP address range directly to the asterisk and dedicate bandwidth to it and it alone. The above is certainly possible, but probably not the most cost effective use of total bandwidth. Based on the words provided, the single link bandwidth should be sized to handle the maximum number of voice channels to be used plus a small amount for web traffic. In a way the sharing of bandwidth with QoS would appear to be the better value option but I can't see that the TC QoS can really be up to the task (again partially this is because I can only control the outgoing traffic shaping - there is nothing I can do about the incoming traffic flooding the bandwidth). Help us understand exactly what this "incoming traffic flooding the bandwidth" is suppose to mean. Are you running something else besides web and voip through this link? If not, then what is "flooding" your bandwidth? ___ --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 -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com ___ --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] Bandwidth: to seperate or not to seperate
inline... You said: Help us understand exactly what this incoming traffic flooding the bandwidth is suppose to mean. Are you running something else besides web and voip through this link? If not, then what is flooding your bandwidth? You are right about web page serving not using much incoming bandwidth (good for one-sided QoS management). I was inaccurate by saying hosting webpages - we also have all email traffic and host racked servers for customers of ours too (and I have them on a lower QoS in the Netfilter/Wondershaper setup). Actually, typically, the servers are business customers and probably don't use much bandwidth at all but I can't be sure that one of them would not, at any time, upload data from their mega-high-speed office connection - its a bit of an unknown. The above comments would be of some concern, particularly if some sends an email to the server with a large attachment (or whatever). Given all the other traffic, you're faced with a choice to micro-manage the existing bandwidth, or, do as you mentioned providing two paths. Some time ago, someone on the list suggested a QoS-like app (maybe it was wondershaper, don't remember) that does impact inbound traffic. My understanding is the app delays TCP response packets (from your server to the external user) essentially slowing the inbound flow of traffic. If you think about how TCP functions, an ACK packet is required after approximately three inbound packets acknowledging the receipt of those three packets; if the ACK packet is delayed by xxx milliseconds, it essentially impacts the speed at which incoming packets arrive. No such thing for UDP traffic though. Since web and email traffic uses TCP, that might be something to look into. It's a bit of a bummer that inbound traffic shaping cannot be done - considering that its a data-center setup as opposed to an office/home setup (kind of a so-close-but-so-far type thing). True inbound packet shaping would actually require the sender to prioritize all packets, and assumes every layer-2 and layer-3 device between the sender and your hardware respect QoS settings. That's not going to happen anytime soon, although some Internet providers do in fact respect it. Thinking about it now, before paying money for something we don't need, I should probably try to graph bandwidth usage (which opens a can of worms too - I'm used to MRTG but it would only show a 5 minute average and not instant peaks that would affect VoIP quality - have you ever used any other graphing tools?). You might try STG to graph the usage. Its available everywhere on the Internet and can be set to poll at one second intervals if that's actually necessary. I use it a lot with five or ten second polling to see peaks, etc. Rich ___ --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] Bandwidth: to seperate or not to seperate
Hi everyone, RE: Bandwidth. We have an asterisk server sharing bandwidth with other [web] servers in cabinets that we rent in a large data-center and all is working fine. But I'm concerned that web traffic could affect the VoIP quality (my tests so far haven't showed this [yet!]. Currently I'm running a server with Netfilter (iptables) between all the servers and the Internet with Forward rules and I'm also including a wondershaper type QoS ruleset with TC to priortise the outgoing VoIP traffic (I say outgoing because this is really the only thing I can shape on the connection as far as I can see). My choice, going forward, is to either buy more bandwidth and magically implement better QoS or the other option is to bring in a separate patch cable, with separate bandwidth, and a different IP address range directly to the asterisk and dedicate bandwidth to it and it alone. In a way the sharing of bandwidth with QoS would appear to be the better value option but I can't see that the TC QoS can really be up to the task (again partially this is because I can only control the outgoing traffic shaping - there is nothing I can do about the incoming traffic flooding the bandwidth). But obviously the data center has fully operational bandwidth shaping to their clients (me!) - or bandwidth throttling might be the better description as they limit my bandwidth to what I buy from them so by buying the separate bandwidth I'd just be relying on their bandwidth shaping rather than trying to do it myself. Does anyone out there know the correct answer to this? I'm sure many people must have come up against this before? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com ___ --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] Bandwidth: to seperate or not to seperate
Inline... RE: Bandwidth. We have an asterisk server sharing bandwidth with other [web] servers in cabinets that we rent in a large data-center and all is working fine. But I'm concerned that web traffic could affect the VoIP quality (my tests so far haven't showed this [yet!]. Currently I'm running a server with Netfilter (iptables) between all the servers and the Internet with Forward rules and I'm also including a wondershaper type QoS ruleset with TC to priortise the outgoing VoIP traffic (I say outgoing because this is really the only thing I can shape on the connection as far as I can see). If your web server is oriented around simply serving up static pages with no one uploading data to it, then the majority of the web traffic will be outbound traffic. (eg, user clicks on a link, small amount of inbound traffic to communicate that click, followed by lots of outbound traffic reppreenting the new page(s) to be viewed.) The wondershaper function should prrioritze that mix of traffic just fine. My choice, going forward, is to either buy more bandwidth and magically implement better QoS or the other option is to bring in a separate patch cable, with separate bandwidth, and a different IP address range directly to the asterisk and dedicate bandwidth to it and it alone. The above is certainly possible, but probably not the most cost effective use of total bandwidth. Based on the words provided, the single link bandwidth should be sized to handle the maximum number of voice channels to be used plus a small amount for web traffic. In a way the sharing of bandwidth with QoS would appear to be the better value option but I can't see that the TC QoS can really be up to the task (again partially this is because I can only control the outgoing traffic shaping - there is nothing I can do about the incoming traffic flooding the bandwidth). Help us understand exactly what this incoming traffic flooding the bandwidth is suppose to mean. Are you running something else besides web and voip through this link? If not, then what is flooding your bandwidth? ___ --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] Bandwidth usage for codecs
hi how much bandwidth is used for the following codecs 723 r 5.3 723 r 6.3 723 r 8 what i know so far is the 723 r 5.3 uses 5.3 k up and 5.3k down 723 r 6.3 uses 6.3 k up and 6.3k down 729 r 8 uses 8 k up and 8k down is this correct or is it like the following 723 r 5.3 uses 11 k up and 11k down 723 r 6.3 uses 13 k up and 13k down 729 r 8 uses 16 k up and 16k down if u guy know, please let me know. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Bandwidth usage for codecs
On Monday October 10 2005 08:18, Kanishka Somaratne spake: hi how much bandwidth is used for the following codecs http://www.voip-info.org/wiki/index.php?page=Bandwidth+consumption -- Joey Kelly Minister of the Gospel | Linux Consultant http://joeykelly.net I may have invented it, but Bill made it famous. --- David Bradley, the IBM employee that invented CTRL-ALT-DEL pgpIckap6836R.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] bandwidth cosume - iax
But, when these clients are on the Internet, all the media flow pass through the asterisk server. Is that way that it works? I can't see what is the difference. It's probably because your two clients are behind NAT routers, which unless you do port forwarding forces all traffic to go trough your asterisk box which is on the net. Is there some way of clients behind NAT communicate peer-to-peer? ___ 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] bandwidth cosume - iax
When I'm connected with two clients in the same LAN of the asterisk server using the same codec and the IAX protocol, I have no media passing through the asterisk. But, when these clients are on the Internet, all the media flow pass through the asterisk server. Is that way that it works? I can't see what is the difference. Thank you and I'm so sorry for my poor english. Wendell ___ 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] bandwidth cosume - iax
Wendell Almeida Silva wrote: When I'm connected with two clients in the same LAN of the asterisk server using the same codec and the IAX protocol, I have no media passing through the asterisk. That is because IAX has a SIP-like reinvite mechanism. So by default, whenever it can, asterisk will step out of the media path which is a good thing unless you want to bill for the call. You can use notransfer=yes in your iax.conf to change this behavior. But, when these clients are on the Internet, all the media flow pass through the asterisk server. Is that way that it works? I can't see what is the difference. It's probably because your two clients are behind NAT routers, which unless you do port forwarding forces all traffic to go trough your asterisk box which is on the net. ___ 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] bandwidth of gsm and g729
what are the bandwidths of the gsm codec and g729 codec and are they in same sound quality . Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] bandwidth og gsm and g729
what are the bandwidths of the gsm codec and g729 codec and are they in same sound quality . __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Bandwidth Reduction using Compressed RTP
Hello: I read many documents about reducing the codec bandwidth by 1)compressing the rtp header and 2)implementing point-to-point link. But none of these documents mentioned how to implement it. So I wonder why there is not much resources about something valuable like this which interest many people, or I just don't see it. I am hoping someone can help. this is one of the many resources I read: http://www.newport-networks.com/whitepapers/voip-bandwidth1.html Thanks; __ Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides! http://travel.yahoo.com/p-travelguide ___ 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] bandwidth
how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. B ___ 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] bandwidth
The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. B ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005 ___ 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] bandwidth
can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. B William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. B ___ 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] bandwidth
Bernie [EMAIL PROTECTED] writes: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. B William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. It can be reduced. Just goole for 'asterisk codecs bandwidth' and click the top link. Regards, Bruno. ___ 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] bandwidth
I think the easiest and most appropriate answer to this is - G729. Later, PauLH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruno Hertz Sent: Tuesday, 5 April 2005 10:21 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] bandwidth Bernie [EMAIL PROTECTED] writes: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. B William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. It can be reduced. Just goole for 'asterisk codecs bandwidth' and click the top link. Regards, Bruno. ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] bandwidth
We use ulaw where we can and g729 where necessary. I think it is like 8k for g729. On Mon, 04 Apr 2005 19:07:24 -0500 Bernie [EMAIL PROTECTED] wrote: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. B William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. B ___ 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 ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ 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] bandwidth
Actually about 80k-82k when you take into account UDP and RTP overhead and assume you are using SIP. Single IAX2 call may be a little less. multiple IAX2 calls using trunking will be a lot less. In fact, this question is answered on http://www.digium.com/index.php?menu=documentation specifically the link to http://www.packetizer.com/voip/diagnostics/bandcalc.html Unfortunatly the above URL is not terribly clear and understandable. People complain about Asterisk's lack of good, organized, understandable documentation. It might help if they actually used the documentation and links that ARE available. Here we have an example of one person that didn't do the research (understandable, since he/she might not have known about the Documentation link on Digium's web site) and then asked a question and then another person that ALSO didn't do the research (I'm guilty of this too, but am getting much better) but answered the question anyway. William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. ___ 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] bandwidth
Bernie wrote: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. Yes. Read up on the various codecs and how much bandwidth they use. ___ 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] Bandwidth
asterisk wrote: Assuming I'm using a VOIP provider of some sort, what kind of bandwidth requirements / line should I expect to have in place? I currently have 8 traditional voice lines, and a FAX line that doubles as my DSL source. Ballpark, what do I need to have in place to move everything to asterisk? - I recommend having a dedicated Linux box (I use debian + a couple of ethernet cards) which does Network Bridge + Asterisk + Traffic shaping. - If your bandwith is short (for example 256kbit/s), install another asterisk box on a dedicated hosting facility with plenty of bandwith. Then buy some g.729 licenses so that you can use g.729 between your office behind DSL and your dedicated box. - Keep your FAX line for faxes, emergencies, and failover. 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
[Asterisk-Users] Bandwidth
Assuming I'm using a VOIP provider of some sort, what kind of bandwidth requirements / line should I expect to have in place? I currently have 8 traditional voice lines, and a FAX line that doubles as my DSL source. Ballpark, what do I need to have in place to move everything to asterisk? Dunc ___ 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] Bandwidth
Dunc Depends on the codec: http://www.voip-info.org/wiki-Bandwidth+consumption Offhand, I would recommend hanging on to the fax line, and pipe it into the asterisk box as an emergency line. That way, people can still dial 911. But that's just me, others will have better ideas I'm sure. Sean asterisk wrote: Assuming I'm using a VOIP provider of some sort, what kind of bandwidth requirements / line should I expect to have in place? I currently have 8 traditional voice lines, and a FAX line that doubles as my DSL source. Ballpark, what do I need to have in place to move everything to asterisk? Dunc ___ ___ 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] Bandwidth, again, can someone check my math?
Bump -- anyone? -Original Message- From: Jay Milk [mailto:[EMAIL PROTECTED] Sent: Friday, January 21, 2005 11:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Bandwidth, again, can someone check my math? I want to put a single voice-mail box on a remote server, where I have metered bandwidth. Before I do this, I want to make sure it's feasible. Could someone confirm the following math for me? G.711, at 64kpbs has a rated network load of 88kbps. So for each second of conversation, about 11KB are crossing the wires in each direction. That means for a minute of two-way conversation, 1.3MB of data are transferred? That means for each GB of bandwith, callers can leave almost 800 minutes worth of voice-messages? Of course, this gets much better if we can get incoming calls on GSM, arriving at something like 2,500 minutes/GB. Is that correct, or did I mess up a decimal point somewhere? ___ 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] Bandwidth, again, can someone check my math?
On January 21, 2005 12:26 pm, Jay Milk wrote: G.711, at 64kpbs has a rated network load of 88kbps. So for each second of conversation, about 11KB are crossing the wires in each direction. 88kbps = 88*1024 bps / 8 bits/byte =11kB/sec, yes, in each direction. That means for a minute of two-way conversation, 1.3MB of data are transferred? Yes, if you take the transmit and receive streams separate. 660kB in each direction. That means for each GB of bandwith, callers can leave almost 800 minutes worth of voice-messages? Seems right to me. 1024*1024 kBytes / 1320kB/min = 794.4 minutes Of course, this gets much better if we can get incoming calls on GSM, arriving at something like 2,500 minutes/GB. And even better if you can get VAD support into * so that it isn't sending back 660kB of silence per minute. Is that correct, or did I mess up a decimal point somewhere? Seems right. -A. ___ 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] Bandwidth, again, can someone check my math?
I want to put a single voice-mail box on a remote server, where I have metered bandwidth. Before I do this, I want to make sure it's feasible. Could someone confirm the following math for me? G.711, at 64kpbs has a rated network load of 88kbps. So for each second of conversation, about 11KB are crossing the wires in each direction. That means for a minute of two-way conversation, 1.3MB of data are transferred? That means for each GB of bandwith, callers can leave almost 800 minutes worth of voice-messages? Of course, this gets much better if we can get incoming calls on GSM, arriving at something like 2,500 minutes/GB. Is that correct, or did I mess up a decimal point somewhere? ___ 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] Bandwidth, computer power
I want to use MY asterisk server to help some people to get MULTIPLE gateways to their VoIP phone. E.g.UK, US, Canada DIDs, registered to my asterisk server, with dedicated dialing to my friend. I believe that the RTP (voice streams) are going directly from the DIDs to my friends, and will not use up my bandwidth. I believe that however, if he use my mailbox it will use my bandwidt, as well he will use twice the bandwidth, if he use a conference call with a third party. Besides the confirmation about the above, I would like how much bandwidth I will need to handle such calls for my friend? bye Ronald ___ 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] Bandwidth Load Balancing / Dundi
Just wondering what other kinds of solutions people have considered/implemented for load balancing bandwidth and IAX connections over the net. Ideas? Results? Suggestions? Experience? -- Kenneth Shaw [EMAIL PROTECTED] ExpiTrans, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth Load Balancing / Dundi
App_conference worked well for me, but after upgading to 1.0.2 this evening, it would no longer compile. I will take a closer look at it soon. But it was much better then app_meetme the most important thing being AGC. (automatic gain control) -Original Message- From: Kenneth Shaw [mailto:[EMAIL PROTECTED] Sent: October 26, 2004 5:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bandwidth Load Balancing / Dundi Just wondering what other kinds of solutions people have considered/implemented for load balancing bandwidth and IAX connections over the net. Ideas? Results? Suggestions? Experience? -- Kenneth Shaw [EMAIL PROTECTED] ExpiTrans, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Bandwidth Load Balancing / Dundi
Correction, with the real 1.0.2 (not the .9 or whatever got accidentally released) It works fine. Check it out, look it up on the wiki. Donny -Original Message- From: Donny Kavanagh Sent: October 27, 2004 1:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Bandwidth Load Balancing / Dundi App_conference worked well for me, but after upgading to 1.0.2 this evening, it would no longer compile. I will take a closer look at it soon. But it was much better then app_meetme the most important thing being AGC. (automatic gain control) -Original Message- From: Kenneth Shaw [mailto:[EMAIL PROTECTED] Sent: October 26, 2004 5:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bandwidth Load Balancing / Dundi Just wondering what other kinds of solutions people have considered/implemented for load balancing bandwidth and IAX connections over the net. Ideas? Results? Suggestions? Experience? -- Kenneth Shaw [EMAIL PROTECTED] ExpiTrans, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Bandwidth control on a home office network
- Original Message - From: Adam Holt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Hi, I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] Thanks for any tips. BR /adam. I use IPCOP - it's another open source project. It does traffic shaping, routing, firewalling, DMZ, etc. It's free and runs on an old PC (I use Pentium 200MHZ w/128MB RAM - but I need it that fast because it's also a content filter for my home network/kids. www.ipcop.org Did I mention, it's free? Roger ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth control on a home office network
Looks like it's time to add a WIKI page on QOS routing alternatives, listing options such as the Linksys WRT (with OpenWRT or Sveasoft or...), m0n0wall, LEAF, etc. It seems that this would be a bit off-topic, but QOS if very much a concern for VOIP. Any volunteers who'd actually know what they're talking about? I'm currently in the research phase of my next router-solution, since it's good-bye for my trusted 5861 soon. -Original Message- From: Roger Hanson [mailto:[EMAIL PROTECTED] Sent: Sunday, October 17, 2004 1:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bandwidth control on a home office network - Original Message - From: Adam Holt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Hi, I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] Thanks for any tips. BR /adam. I use IPCOP - it's another open source project. It does traffic shaping, routing, firewalling, DMZ, etc. It's free and runs on an old PC (I use Pentium 200MHZ w/128MB RAM - but I need it that fast because it's also a content filter for my home network/kids. www.ipcop.org Did I mention, it's free? Roger ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Bandwidth control on a home office network
Jay Milk wrote: Looks like it's time to add a WIKI page on QOS routing alternatives, listing options such as the Linksys WRT (with OpenWRT or Sveasoft or...), m0n0wall, LEAF, etc. It seems that this would be a bit off-topic, but QOS if very much a concern for VOIP. Any volunteers who'd actually know what they're talking about? I'm currently in the research phase of my next router-solution, since it's good-bye for my trusted 5861 soon. Qos in general http://www.voip-info.org/tiki-index.php?page=QoS Small/Home Routers with QoS http://www.voip-info.org/wiki-VOIP+Routers Please add information! Thanks. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth control on a home office network
Hello: Try ipfw and dummynet on a freebsd box acting as traffic shapper, you can put your ATAs or like in another network, then you can manage your bandwith like as your ISP. If you need some assistance don't hessitate ask me Excuse me by the offtopic --- Ing. Julio Alvarez Tejera Unix Trends *BSD, Solaris Linux CT VoIP Solutions Finder (506) 286-5478 +1-305-704-2019 --- estabilidad al extremo - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, October 16, 2004 12:25 PM Subject: Re: [Asterisk-Users] Bandwidth control on a home office network I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] QoS, regardless of whether its based on the IP header TOS bits or on specific tcp/udp port numbers, essentially prioritizes the outbound flow of data packets, sending high priority packets before lower priority packets. It does nothing for inbound data such as downloads to your site. Most broadband connections have a different upload vs download speed, where usually the download speed is substantially greater then the upload speed. E.g., not uncommon to see DSL or Cable modems limited to 758k down and 128k/256k upload speeds. QoS may help with prioritizing traffic through the 128k/256k. However, your internet service provider would need to prioritize the download traffic for you. There are some rather expensive devices that you can install that will rate limit both upload and download traffic. Those devices artifically control the download traffic by withholding TCP acknowledgment packets, etc. Not sure how effective they are though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Bandwidth control on a home office network
On Sat, 16 Oct 2004 21:00:31 +0200, gramels wrote: you might consider http://m0n0.ch/wall on a soekris.com or pcengines.ch board which does nice trafficshaping for little money. m0n0wall is a freebsd based opensource firewall appliance I heartily concur! I used m0n0wall on a Soekris 4501 to replace a Linksys BEFSR-81. m0n0 is a joy to use. You can try the PC version that ony requires a dual NIC'd old PC as a testbed to get you started. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Stay calm. Be brave. Watch for the signs. - Anne Sloboda, on the occasion of my wedding ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth control on a home office network
Hi, I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] Thanks for any tips. BR /adam. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth control on a home office network
Adam Holt wrote: Hi, I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] Thanks for any tips. BR /adam. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Cisco 7960 does QoS by setting TOS bits. Furthermore, it works best when used with other Cisco switches (especially ones with the new AutoQoS feature). You need something that can understand those, back to your original router problem. If you are in the US, you can pickup a Linksys WRT54GS for well under $100 that you can do an amazing amount of things on. You can now even do QoS with the Linksys firmware, although alternative firmware (Sveasoft, OpenWRT, etc.) is more fun. A cisco router would work, but I don't think that you need to blow that much money for a home router, especially when the Linksys WRT's are so much fun! I actually don't know if any of the LinkSys firmware understands ToS, but you can do Qos and bandwidth shaping using other means. Check it out. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth control on a home office network
I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] QoS, regardless of whether its based on the IP header TOS bits or on specific tcp/udp port numbers, essentially prioritizes the outbound flow of data packets, sending high priority packets before lower priority packets. It does nothing for inbound data such as downloads to your site. Most broadband connections have a different upload vs download speed, where usually the download speed is substantially greater then the upload speed. E.g., not uncommon to see DSL or Cable modems limited to 758k down and 128k/256k upload speeds. QoS may help with prioritizing traffic through the 128k/256k. However, your internet service provider would need to prioritize the download traffic for you. There are some rather expensive devices that you can install that will rate limit both upload and download traffic. Those devices artifically control the download traffic by withholding TCP acknowledgment packets, etc. Not sure how effective they are though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth control on a home office network
you might consider http://m0n0.ch/wall on a soekris.com or pcengines.ch board which does nice trafficshaping for little money. m0n0wall is a freebsd based opensource firewall appliance On Sat, 16 Oct 2004 12:25:06 -0600, Rich Adamson [EMAIL PROTECTED] wrote: I have a Grandstream ATA today connected to my 750k broadband connection via an older router / firewall that doesn't have any QoS / ToS capability. It works fine apart from the obvious problem of when large emails come in or somebody else on the network starts d/l-ing something big off the web. I'm wondering whether to swap the router for a Cisco in order to introduce some local bandwidth control. Alternatively I was wondering if I picked up a Cisco 7960 handset instead - is the 2nd ethernet port routed through the device, or does it just act as an Ethernet repeater, i.e. if I arranged the handset in the network as below would I get bandwidth prioritisation for the 7960? [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY NETWORK] QoS, regardless of whether its based on the IP header TOS bits or on specific tcp/udp port numbers, essentially prioritizes the outbound flow of data packets, sending high priority packets before lower priority packets. It does nothing for inbound data such as downloads to your site. Most broadband connections have a different upload vs download speed, where usually the download speed is substantially greater then the upload speed. E.g., not uncommon to see DSL or Cable modems limited to 758k down and 128k/256k upload speeds. QoS may help with prioritizing traffic through the 128k/256k. However, your internet service provider would need to prioritize the download traffic for you. There are some rather expensive devices that you can install that will rate limit both upload and download traffic. Those devices artifically control the download traffic by withholding TCP acknowledgment packets, etc. Not sure how effective they are though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Bandwidth requirement with G729A
I believe that is due to a packet sizing issue. The g.729 codec likes to make a whole bunch of small packets to reduce voice latency. This has the unfortunate side effect of increasing overhead to a level that is above what you would think. More packets=more overhead On 13 Jul 2004 15:36:28 -0400, Sudhir Kumar [EMAIL PROTECTED] wrote: I am using G729A. I was expecting a maximum bandwith requirement of 24Kbps (8Kbps for voice, 16Kbps for IP overhead), however when I measure the actual usage, it comes around 30 to 32 Kbps. What happens? Thanks, -- sudhir ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Bandwidth requirement with G729A
Sudhir Kumar [EMAIL PROTECTED] wrote: I am using G729A. I was expecting a maximum bandwith requirement of 24Kbps (8Kbps for voice, 16Kbps for IP overhead), however when I measure the actual usage, it comes around 30 to 32 Kbps. What happens? The 8k is each way, so that makes 16k for voice. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Bandwidth requirement with G729A
On 08:56 AM 7/14/2004, Kevin Walsh wrote: The 8k is each way, so that makes 16k for voice. Huh? I'm not sure the relevance of this as it applies to the original question. If the person asking the original question had said that they were looking at adding both inbound and outbound traffic for a total, then yes. However, I think they were only looking at one way traffic since when most people talk about traffic they speak of one direction (and in many cases in the network world, since your bandwidth is generally identical in capacity for both directions, the largest usage is generally the discussed value). In other words, I don't say my T1 is 3.088 Mbps because it's 1.544 in and 1.544 out. However in the case of ADSL I most often refer to it in a manner denoting both the inbound and outbound (256/2048 for example). And in the case of measurement, most measurement tools display bandwidth usage for each direction separately, otherwise troubleshooting would be severely crippled. So the issue the original poser is having is that they are unable to account for measured traffic as opposed to signal traffic. signal is 8kbps but then you have the overhead associated with the UDP and IP wrappers as well as the overhead associated with the ethernet wrapper. There was a post a while back where someone had gone through and listed the overhead specifically per 'wrapper' at least down to the ethernet frame level. You should be able to google it up for specifics. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
