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
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 > 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
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 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
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
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
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 > mgravesmstvp.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
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
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 mgravesmstvp.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
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
[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
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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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
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 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
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
"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
[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
[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
