I am using a VoIP Router, which can manage QoS etc...
The only problem, that it cannot decrease a upload, if I receive a VoIP
phone call.....
I need to limit the bandwidth of my upload permanently (for the Website,
like max 4*32kbp )
I would expect to find one, which can decrease itself the upload, IF a
phone call is arriving !
Eh ? The purpose of QoS is to be able to give priority to certain type of
packets, in this case VoIP ones.
Therefore ANY router with QoS should be able to limit downloads when VoIP is
active.
Regards,
Martin
Martin² wrote:
> Eh ? The purpose of QoS is to be able to give priority to certain type of
> packets, in this case VoIP ones.
> Therefore ANY router with QoS should be able to limit downloads when VoIP is
> active.
> Regards,
> Martin
>
>
Hi Martin,
Maybe I was not really clear :-(
Suppose that somebody is downloading a big ISO file from your website.
(your website is located in your house)
=> the upload rate is maximum.....
Now, a phone call is arriving. for me, a clever router, will decrease
the upload (from your website) in order to be let you make your phone
call...
And this is precisely, what I cannot find. I need to give permanently
80% of my upload bandwidth to my website... in order to make phone
calls... with a correct audible voice.
"Steve" <steve@everybody.com> wrote in message
news:fn9jc2$fs3$1$8302bc10@news.demon.co.uk
[snip]
: : Hi Martin,
: :
: : Maybe I was not really clear :-(
: :
: : Suppose that somebody is downloading a big ISO file
: : from your website. (your website is located in your
: : house) => the upload rate is maximum.....
: :
: : Now, a phone call is arriving. for me, a clever router,
: : will decrease the upload (from your website) in order
: : to be let you make your phone call...
: : And this is precisely, what I cannot find. I need to
: : give permanently 80% of my upload bandwidth to my
: : website... in order to make phone calls... with a
: : correct audible voice.
For this amongst other reasons, it's not really a good idea to host a
website at home. Far better to use a commercial hosting service.
In article <fn9jc2$fs3$1$8302bc10@news.demon.co.uk>,
Steve <steve@everybody.com> wrote:
>Martin² wrote:
>> Eh ? The purpose of QoS is to be able to give priority to certain type of
>> packets, in this case VoIP ones.
>> Therefore ANY router with QoS should be able to limit downloads when VoIP is
>> active.
>> Regards,
>> Martin
>>
>>
>
>Hi Martin,
>
>Maybe I was not really clear :-(
>
>Suppose that somebody is downloading a big ISO file from your website.
>(your website is located in your house)
>=> the upload rate is maximum.....
>
>Now, a phone call is arriving. for me, a clever router, will decrease
>the upload (from your website) in order to be let you make your phone
>call...
>And this is precisely, what I cannot find. I need to give permanently
>80% of my upload bandwidth to my website... in order to make phone
>calls... with a correct audible voice.
You're far better off as Martin suggests and have a router that will
prioritise certian data over others. That way someone can get 100% of
your bandwidth when getting a file, but any VoIP data will have priority
over web (or other) data.
Drayteks come close to being able to do this - the one I have, (an older
2600, flashed with the latest firmware) seems to cope just fine when
I'm pushing many MB of data to remote servers at the same time as making
VoIP calls.
Eg. an SCP of a 1MB file to one of my remote servers:
Steve,
the Draytek 2600VG I am using has an option to use QoS in either or BOTH
directions, and I ofcourse set it to BOTH for VoIP.
The other VoIP router I am familiar with, Zyxel 2602HWL does (IIRC) the
same.
I would have thought any QoS router would work in both directions.
Which one are you using ?
Regards,
Martin
On Thu, 24 Jan 2008 09:31:56 +0000 (UTC), Gordon Henderson
<gordon+usenet@drogon.net> wrote:
>Drayteks come close to being able to do this - the one I have, (an older
>2600, flashed with the latest firmware) seems to cope just fine
My 2600 has the Long line firmware - v2.5.6_UK2
and I can't recall ever having seen any QoS settings.
Where are they please?
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On Wed, 23 Jan 2008 19:32:30 +0000, Steve <steve@everybody.com> wrote:
>Hi,
>
>I am using a VoIP Router, which can manage QoS etc...
>The only problem, that it cannot decrease a upload, if I receive a VoIP
>phone call.....
I've read somewhere that for VoIP trunk calls (i.e. not internal) QoS
doesn't make a lot of difference unless your Broadband provider also
honours QoS.
But I'd like to be corrected!
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In article <d55jp35c0hpjj4ub555kmbq6dsel5ki8u0@4ax.com>,
David Quinton <usenet_2005D_email@REMOVETHISBITbizorg.co.uk> wrote:
>On Thu, 24 Jan 2008 09:31:56 +0000 (UTC), Gordon Henderson
><gordon+usenet@drogon.net> wrote:
>>Drayteks come close to being able to do this - the one I have, (an older
>>2600, flashed with the latest firmware) seems to cope just fine
>
>My 2600 has the Long line firmware - v2.5.6_UK2
>and I can't recall ever having seen any QoS settings.
>
>Where are they please?
In article <gd5jp31k4jj23lr2mul32i9u0uaf0i6939@4ax.com>,
David Quinton <usenet_2005D_email@REMOVETHISBITbizorg.co.uk> wrote:
>On Wed, 23 Jan 2008 19:32:30 +0000, Steve <steve@everybody.com> wrote:
>
>>Hi,
>>
>>I am using a VoIP Router, which can manage QoS etc...
>>The only problem, that it cannot decrease a upload, if I receive a VoIP
>>phone call.....
>
>
>I've read somewhere that for VoIP trunk calls (i.e. not internal) QoS
>doesn't make a lot of difference unless your Broadband provider also
>honours QoS.
>
>But I'd like to be corrected!
You're almost right.
You can apply QoS to outgoing packets as the router can queue up a few
outgoing packets before it clocks them over the wire (the slow part), so
it can make sure VoIP packets are delivered in a timely manner while
delaying lower priority data.
However for incoming packets, once they've been clocked over the wire
and arrive in the router, it's too late, as they've already clocked up
"wire time" to transmit the data.
Of-course on the outgoing side, all you're doing is prioritising your own
outgoing traffic. Once it reaches the ISP, and onto the global Internet
you have no more control over it. Some ISPs (and I've worked with one in
Bristol who can do this on leased lines) will prioritise VoIP traffic
through their own network, for you if you ask them, but then it's only
through their own network to their obrders with other ISP/the global
Internet (and in his case, his quote was that the easiest way to manage
contention was to not have any!)
To manage incoming traffic, there's not a lot you can do. There are some
tricks like sending various TCP control packets to sending sites to try
to get hem to slow down, and there are some routers that will do this
(cue Tim ;-) but I'm not convinced it's really viable, as, again, once
they think a remote site is sending too fast, the damage is done and
you've already had packet loss/delays.
In the real world, if you want good VoIP then you have to be prepared to
pay for a good ISP. Using "domestic" ISPs (especially for a business)
is a false economy IMO. I now have a number of customers using an
Internet (ADSL) connection to make outgoing calls on (and some taking
incoming), and in all cases where they are using an inferior ISP they've
had problems. Move them to a good business quality ISP (and we're only
taking £25 a month here) and things have been much better.
Thanks Gordon.
Mines a plain 2600 - so 2.5.6 is the most recent.
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On Fri, 25 Jan 2008 09:28:22 +0000 (UTC), Gordon Henderson
<gordon+usenet@drogon.net> wrote:
>In the real world, if you want good VoIP then you have to be prepared to
>pay for a good ISP. Using "domestic" ISPs (especially for a business)
>is a false economy IMO. I now have a number of customers using an
>Internet (ADSL) connection to make outgoing calls on (and some taking
>incoming), and in all cases where they are using an inferior ISP they've
>had problems. Move them to a good business quality ISP (and we're only
>taking £25 a month here) and things have been much better.
Interesting, thanks.
I think our Demon Business has 2x the upload speed of the non-business
packages.
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Steve wrote:
> Hi,
>
> I am using a VoIP Router, which can manage QoS etc...
> The only problem, that it cannot decrease a upload, if I receive a VoIP
> phone call.....
> I need to limit the bandwidth of my upload permanently (for the Website,
> like max 4*32kbp )
> I would expect to find one, which can decrease itself the upload, IF a
> phone call is arriving !
>
> - Do you know a clever one, which can do that ?
>
> thanks and regards,
>
> Steve
Just thinking laterally here, have you thought about asking your VoIP
provider to enable an 8k codec for your calls. this reduces your 64k
needed down by a power of 8 and usually allows a maintainable level of
clarity for your VoIP calls...
Also, we recommend the billion router range which allows you to
prioritise all packets and ports in % terms up & down
In article <o0hjp3l4586v0qkqon1c8l982en8rj657r@4ax.com>,
David Quinton <usenet_2005D_email@REMOVETHISBITbizorg.co.uk> wrote:
>On Fri, 25 Jan 2008 09:28:22 +0000 (UTC), Gordon Henderson
><gordon+usenet@drogon.net> wrote:
>
>>In the real world, if you want good VoIP then you have to be prepared to
>>pay for a good ISP. Using "domestic" ISPs (especially for a business)
>>is a false economy IMO. I now have a number of customers using an
>>Internet (ADSL) connection to make outgoing calls on (and some taking
>>incoming), and in all cases where they are using an inferior ISP they've
>>had problems. Move them to a good business quality ISP (and we're only
>>taking £25 a month here) and things have been much better.
>
>Interesting, thanks.
>
>I think our Demon Business has 2x the upload speed of the non-business
>packages.
It's easy to check - just look for some sort of stats page on your
router - you'll either have 256Kbits/sec, up to 448Kbits/sec or
up to 832Kbits/sec depending on the service. The actual speed will
depends on the same factors as download speed - distance, cable quality,
etc.
Then the speed you'll get will depends on the contention in the BT
wholesale network (if Demon use it), then the centention inside Demons
own network, then the quality of their pipes to other ISPs at the
peering points, etc.
In article <4799DF4C.4050205@voicehost.co.uk>,
Phil Collin <phil.collin@voicehost.co.uk> wrote:
>Steve wrote:
>> Hi,
>>
>> I am using a VoIP Router, which can manage QoS etc...
>> The only problem, that it cannot decrease a upload, if I receive a VoIP
>> phone call.....
>> I need to limit the bandwidth of my upload permanently (for the Website,
>> like max 4*32kbp )
>> I would expect to find one, which can decrease itself the upload, IF a
>> phone call is arriving !
>>
>> - Do you know a clever one, which can do that ?
>>
>> thanks and regards,
>>
>> Steve
>
>Just thinking laterally here, have you thought about asking your VoIP
>provider to enable an 8k codec for your calls. this reduces your 64k
>needed down by a power of 8 and usually allows a maintainable level of
>clarity for your VoIP calls...
It's nearer to 80Kb/sec when you add in the RTP and IP overhead to the
data stream...
The 8K one is g729... Which is covered by various software patents. Not
applicable in the UK though ... and i've had customers complain of sound
quality problems when testing it though.
It also needs more cpu to transcode to the outside world at the VoIP
providers end though, although a good server is going to be able to cope
with 100's of calls before there is a problem.
I've used g726 which is half the bandwidth of g711 with good results -
it's what DECT phones use internally AIUI - it has a bit-rate of
32Kb/sec (plus SIP & IP overhead)
>Also, we recommend the billion router range which allows you to
>prioritise all packets and ports in % terms up & down
Intersting. I'm getting "irritated" by some of the newer Drayteks these
days as they seem to have various weird problems to do with NAT tables
breaking...
Phil Collin wrote:
> Steve wrote:
>> Hi,
>>
>> I am using a VoIP Router, which can manage QoS etc...
>> The only problem, that it cannot decrease a upload, if I receive a
>> VoIP phone call.....
>> I need to limit the bandwidth of my upload permanently (for the
>> Website, like max 4*32kbp )
>> I would expect to find one, which can decrease itself the upload, IF a
>> phone call is arriving !
>>
>> - Do you know a clever one, which can do that ?
>>
>> thanks and regards,
>>
>> Steve
>
> Just thinking laterally here, have you thought about asking your VoIP
> provider to enable an 8k codec for your calls. this reduces your 64k
> needed down by a power of 8 and usually allows a maintainable level of
> clarity for your VoIP calls...
>
> Also, we recommend the billion router range which allows you to
> prioritise all packets and ports in % terms up & down
>
> regards,
>
> Phil
Hi,
Thnaks for all these info,
I personally use a Billion BiPAC 7402G, and an Asterisk 1.4.4 server
I have setup ion my SIP.CONF, tos=0xb8
And in my Billion 7402, I selected (Port Setting) the IPv4 TOS Priority
Control, and select 184 (0xb8)
I would say, that the quality is not bad at all.. even using a Webserver
at home... I didn't receive any complaint of bad transmission quality.
I agree that for the coming voice (download), there is no QoS from the
VOIP provider, but my ADSL Speed is quite good enough. My problem is
only on the upload side.
The problem, is that the BiPAC is not clever enough to decrease, the
Upload (on the web) if a call arrive.
>"Andy Burns" <usenet.jan2008@adslpipe.co.uk> wrote in message
news:13pjmo7qit966c6@corp.supernews.com...
>On 25/01/2008 02:48, Martin² wrote:
>> I would have thought any QoS router would work in both directions.
>QoS is easier to apply to packets you are sending out of an interface,
>you have much less control over what packets others are aiming towards you.
actually - you dont have any.
all you can do is tinker with what goes back, and cross fingers that it
affects the return traffic.....
FWIW "real QoS" needs traffic management anywhere packets go into a
bottleneck to make sure the important ones get thru with the right latency /
jitter / loss rates.
so in the worst case with full load traffic both ways, any SoHo router on an
internet feed cannot fully protect a voice conversation unless your ISP does
something similar at their end (and AFAICT none of them do).
--
Regards
"Steve" <steve@everybody.com> wrote in message
news:fndn8v$fgh$1$8300dec7@news.demon.co.uk...
> Phil Collin wrote:
> > Steve wrote:
> >> Hi,
> >>
> >> I am using a VoIP Router, which can manage QoS etc...
> >> The only problem, that it cannot decrease a upload, if I receive a
> >> VoIP phone call.....
> >> I need to limit the bandwidth of my upload permanently (for the
> >> Website, like max 4*32kbp )
> >> I would expect to find one, which can decrease itself the upload, IF a
> >> phone call is arriving !
> >>
> >> - Do you know a clever one, which can do that ?
> >>
> >> thanks and regards,
> >>
> >> Steve
> >
> > Just thinking laterally here, have you thought about asking your VoIP
> > provider to enable an 8k codec for your calls. this reduces your 64k
> > needed down by a power of 8 and usually allows a maintainable level of
> > clarity for your VoIP calls...
> >
> > Also, we recommend the billion router range which allows you to
> > prioritise all packets and ports in % terms up & down
> >
> > regards,
> >
> > Phil
>
>
> Hi,
>
> Thnaks for all these info,
>
> I personally use a Billion BiPAC 7402G, and an Asterisk 1.4.4 server
> I have setup ion my SIP.CONF, tos=0xb8
> And in my Billion 7402, I selected (Port Setting) the IPv4 TOS Priority
> Control, and select 184 (0xb8)
>
> I would say, that the quality is not bad at all.. even using a Webserver
> at home... I didn't receive any complaint of bad transmission quality.
> I agree that for the coming voice (download), there is no QoS from the
> VOIP provider, but my ADSL Speed is quite good enough. My problem is
> only on the upload side.
> The problem, is that the BiPAC is not clever enough to decrease, the
> Upload (on the web) if a call arrive.
given the cost of ADSL - if you have enough traffic to justify a A** server,
then why not give it a separate ADSL link?
No contention with other traffic = no problem with QoS.....
>
> I need to limit the Webserver
>
> Regards,
>
> Steve
>
--
Regards
>> "Andy Burns" <usenet.jan2008@adslpipe.co.uk> wrote in message
>
>> you have much less control over what packets others are aiming towards you.
>
> actually - you dont have any.
>
> all you can do is tinker with what goes back, and cross fingers that it
> affects the return traffic.....
yes, as I said, much less control (where less may include none) SYNs are
difficult to control, other than with ICMP source quench, but if it's an
established connection you can shut down the receive window.
> so in the worst case with full load traffic both ways, any SoHo router on an
> internet feed cannot fully protect a voice conversation unless your ISP does
> something similar at their end (and AFAICT none of them do).
It works pretty well for sites on private IPstream networks where remote
sites have ADSL max and a 34Mbps ATM "central" arrives at HQ over ATM
though.
Gordon:
>In the real world, if you want good VoIP then you have to be prepared to
>pay for a good ISP.
I am on the cheapest deal available, PlusNet at £9.99,
my VoIP works just fine (with 711 codec) since I set up QoS on my Draytek.
Mind you PlusNet does traffic shaping which supposedly gives priority to
VoIP.
Regards,
Martin
On Fri, 25 Jan 2008 15:33:31 +0000 (UTC), Gordon Henderson
<gordon+usenet@drogon.net> wrote:
>It's easy to check - just look for some sort of stats page on your
>router - you'll either have 256Kbits/sec, up to 448Kbits/sec or
>up to 832Kbits/sec depending on the service. The actual speed will
>depends on the same factors as download speed - distance, cable quality,
>etc.
Router says:
Up - 640000
Down - 2528000
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"Andy Burns" <usenet.jan2008@adslpipe.co.uk> wrote in message
news:13pku3slsv3brfd@corp.supernews.com...
> On 25/01/2008 23:38, stephen wrote:
>
> >> "Andy Burns" <usenet.jan2008@adslpipe.co.uk> wrote in message
> >
> >> you have much less control over what packets others are aiming towards
you.
> >
> > actually - you dont have any.
> >
> > all you can do is tinker with what goes back, and cross fingers that it
> > affects the return traffic.....
>
> yes, as I said, much less control (where less may include none) SYNs are
> difficult to control, other than with ICMP source quench, but if it's an
> established connection you can shut down the receive window.
>
> > so in the worst case with full load traffic both ways, any SoHo router
on an
> > internet feed cannot fully protect a voice conversation unless your ISP
does
> > something similar at their end (and AFAICT none of them do).
>
> It works pretty well for sites on private IPstream networks where remote
> sites have ADSL max and a 34Mbps ATM "central" arrives at HQ over ATM
> though.
it might do when you test it - as long as there isnt a bottle neck where the
traffic gets to the DSLAM, and where the DSLAM sends into the ADSL line.
if you manage to congest either of those points enough to cause delay or
loss of the voice then you are scuppered, unless someone gives your traffic
priority (some might, but i havent come across an ISP who claims to do it)
The DSLAM link isnt even visible to you since you share it with all users on
the DSLAM and the set of ISPs that share that plumbing - but an overload and
no QoS is going to hurt.
Your ADSL link is more under your control so to test it, try downloading
from a news server or an ISO distie so that your inbound line is saturated.
Then start a voice call and see what happens.....
>
--
Regards
> it might do when you test it - as long as there isnt a bottle neck where the
> traffic gets to the DSLAM, and where the DSLAM sends into the ADSL line.
Agreed that contention could kill it in theory at least, but this is
with IPstream Max Premium, so at least the non-Premium users will start
complaining first first if any particular exchange gets busy.
> if you manage to congest either of those points enough to cause delay or
> loss of the voice then you are scuppered, unless someone gives your traffic
> priority (some might, but i havent come across an ISP who claims to do it)
True, did ask BT for QoS but they laughed.
> Your ADSL link is more under your control so to test it, try downloading
> from a news server or an ISO distie so that your inbound line is saturated.
The ADSL connections from the remote sites do not connect to the
internet, they connect as ATM to the central HQ site, have done tests
sending/receiving large files at same time as priority traffic, works
nicely with CBWFQ at each end.
stephen wrote:
> "Steve" <steve@everybody.com> wrote in message
> news:fndn8v$fgh$1$8300dec7@news.demon.co.uk...
>> Phil Collin wrote:
>>> Steve wrote:
>>>> Hi,
>>>>
>>>> I am using a VoIP Router, which can manage QoS etc...
>>>> The only problem, that it cannot decrease a upload, if I receive a
>>>> VoIP phone call.....
>>>> I need to limit the bandwidth of my upload permanently (for the
>>>> Website, like max 4*32kbp )
>>>> I would expect to find one, which can decrease itself the upload, IF a
>>>> phone call is arriving !
>>>>
>>>> - Do you know a clever one, which can do that ?
>>>>
>>>> thanks and regards,
>>>>
>>>> Steve
>>> Just thinking laterally here, have you thought about asking your VoIP
>>> provider to enable an 8k codec for your calls. this reduces your 64k
>>> needed down by a power of 8 and usually allows a maintainable level of
>>> clarity for your VoIP calls...
>>>
>>> Also, we recommend the billion router range which allows you to
>>> prioritise all packets and ports in % terms up & down
>>>
>>> regards,
>>>
>>> Phil
>>
>> Hi,
>>
>> Thnaks for all these info,
>>
>> I personally use a Billion BiPAC 7402G, and an Asterisk 1.4.4 server
>> I have setup ion my SIP.CONF, tos=0xb8
>> And in my Billion 7402, I selected (Port Setting) the IPv4 TOS Priority
>> Control, and select 184 (0xb8)
>>
>> I would say, that the quality is not bad at all.. even using a Webserver
>> at home... I didn't receive any complaint of bad transmission quality.
>> I agree that for the coming voice (download), there is no QoS from the
>> VOIP provider, but my ADSL Speed is quite good enough. My problem is
>> only on the upload side.
>> The problem, is that the BiPAC is not clever enough to decrease, the
>> Upload (on the web) if a call arrive.
>
> given the cost of ADSL - if you have enough traffic to justify a A** server,
> then why not give it a separate ADSL link?
>
> No contention with other traffic = no problem with QoS.....
>> I need to limit the Webserver
>>
>> Regards,
>>
>> Steve
>>
Hi Stephen,
No. I won't say that I have a lot of traffic. I was quite happy to
install a Asterisk server. That gave me some flexibility, but if I could
manage and guaranty a perfect phone conversation ( and I think that I
would be able to do it with my ADSL bandwidth and one ADSL line, that
would be great.
Steve wrote:
>
> Suppose that somebody is downloading a big ISO file from your website.
> (your website is located in your house)
> => the upload rate is maximum.....
>
> Now, a phone call is arriving. for me, a clever router, will decrease
> the upload (from your website) in order to be let you make your phone
> call...
> And this is precisely, what I cannot find. I need to give permanently
> 80% of my upload bandwidth to my website... in order to make phone
> calls... with a correct audible voice.
>
You need an CTX-1000 from Converged access. ProVu sell them in the UK.
Phil Collin wrote:
>
> Just thinking laterally here, have you thought about asking your VoIP
> provider to enable an 8k codec for your calls. this reduces your 64k
> needed down by a power of 8 and usually allows a maintainable level of
> clarity for your VoIP calls...
Depends if you can cope with the calls. G.723 gives me headaches after
about 30 seconds of calls.
Also, you don't really save that much bandwidth. By the time you've
added on RTP headers, UDP headers, IP headers, PPP overhead, split it
into 53 byte ATM frames ..
You end up with much more overhead than voice on the lower rate codecs.
> Also, we recommend the billion router range which allows you to
> prioritise all packets and ports in % terms up & down
But are the priorities dynamic? Do you get full throughput when no
VoIP calls are happening?
Andy Burns wrote:
> On 25/01/2008 02:48, Martin² wrote:
>
>> I would have thought any QoS router would work in both directions.
>
> QoS is easier to apply to packets you are sending out of an interface,
> you have much less control over what packets others are aiming towards you.
>
Yes, but by adding latency to certain slows, and slowing down (but not
dropping) acknowledgements in the opposite direction, then you can get a
serviceable solution.
Most `office` traffic is TCP.
If somebody decides to dump traffic at your IP address, then there is
absolutely nothing you can do about it at the receiving end. (well, you
could phone your ISP and ask them to firewall it - not a practical solution.
Not that I'd want to part with my hard-earned to anyone who uses
phrases such as:-
"Enhances your VoIP experience"...
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"David Quinton"
<usenet_2005D_email@REMOVETHISBITbizorg.co.uk> wrote in
message news:i8ntp3hgjh3kl15va37v2cc565s1q968r6@4ax.com
: : On Mon, 28 Jan 2008 16:14:39 +0000, Tim
: : <nutnews@kooky.org> wrote:
: :
: : :
: : : You need an CTX-1000 from Converged access. ProVu
: : : sell them in the UK.
: :
: : <http://www.provu.co.uk/converged_ctx1000.html>
: : £350
: :
: : Not that I'd want to part with my hard-earned to anyone
: : who uses phrases such as:-
: : "Enhances your VoIP experience"...
I know the feeling. I keep wanting to tell shop staff I don't *want* a
"shopping experience", I just want to buy some groceries..!
BTW when did shops become "stores"..? ("wonderful offers in store" signs
all over the windows)
Ivor
QoS Router 
Linear Mode
