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Voip implementation

 
 
Trilok
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Posts: n/a
 
      04-21-2004
Hi,

I am an IT graduate student and doing a term paper on Voip
implementation. I am a assuming a company XYZ has a point to point T1
connection between three locations(Cincinnati,Chicago & Detroit).The
company plans to implement Voip at each of these locations.There are
fifty users at each of the location.My questions are:

1.How many phone & fax lines can be accomodated on the T1 line?
2.Will the T1 line be able to handle all 50 users simultaneously?
3.How do you calculate the above(1) & (2)?
4.What is point-to-point T1 line: does it mean that there is a
dedicated line that runs between one location's Lan to the
other location's Lan?
or there is a dedicated line between one location & the internet
backbone (ib) & from the ib to the other location?
5.How does one go about figuring out the cost of
hardware,software,manpower & time to implement Voip in the company
XYZ?

Would really appreciate if you can let me know on the above.

thanks,
Teju
 
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Walter Roberson
Guest
Posts: n/a
 
      04-21-2004
In article <(E-Mail Removed) >,
Trilok <(E-Mail Removed)> wrote:
: I am an IT graduate student and doing a term paper on Voip
:implementation.

: 1.How many phone & fax lines can be accomodated on the T1 line?

A channelized T1 would have 24 independant timeslots, 1.544 megabits
per second total. Each timeslot could be used to carry a telephone-
company quality call of 8000 samples per second, 8 bits per sample.
(The extra 8000 bits are used to carry control information.)

That's for standard calls. VOIP would, though, typically use IP
and compression techniques to reduce the data stream requirements,
and for VOIP you wouldn't necessarily want to channelize your T1.

: 2.Will the T1 line be able to handle all 50 users simultaneously?

You can always adjust the VOIP lossy compression algorithm parameters
until the data fits. You need a minimum-quality metric in order to
make a decision about how many VOIP can be carried.

: 3.How do you calculate the above(1) & (2)?

(1) is by specification of T1, which you can research in IETF
standards (or just look up on some page or other at cisco.com)

(2) is the much more difficult question, as it depends upon the
quality of your perceptual coding algorithms and upon your standards
of intelligability at the other end. It also depends on whether the
VOIP is truly being used to carry -voice-, or if sometimes you want
to run fax over it, or if sometimes you want high-quality music...


: 4.What is point-to-point T1 line: does it mean that there is a
: dedicated line that runs between one location's Lan to the
: other location's Lan?

Yup, pretty much. It might go through some switching equipment at
various telco's along the way, but there would be a dedicated circuit
(and probably a dedicated timeslot) on each and every one of those
switches)

: or there is a dedicated line between one location & the internet
: backbone (ib) & from the ib to the other location?

Not for a point-to-point line. But there are variations of that
approach such as ATM in which what one gets is dedicated virtual
circuits. Point-to-point T1's always connect the same two locations;
virtual circuits in some of the other technologies allow bandwidth
guarantees to be established for the duration of a session, with
the endpoints being determinable dynamically (provided the endpoints
are both in the service area of the technology.)

: 5.How does one go about figuring out the cost of
:hardware,software,manpower & time to implement Voip in the company
:XYZ?

One hires a consultant who has done it before. If you try to do
VOIP with people who are unfamiliar with the technology and haven't
had to do similar real-time work before, chances are excellent that
mistakes will be made, configurations will be experimented with,
the wrong equipment will be bought, the wrong dedicated line type will
be installed (on a multi-year contract), the internal switches won't
be upgraded to QoS, or won't be upgraded to handle enough simultaneous
channels... etc., etc.. So if you aren't hiring a consultant to
determine all these prices on your behalf, then whatever figure you
come up with yourself, you had better multiply by about 8 for hardware
and change the implimentation time to "person-years" where you had
"person-months" before.
--
Are we *there* yet??
 
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Joe Technician
Guest
Posts: n/a
 
      04-21-2004
Questions questions questions.

The point to point T1 for Data would just be a 1.544M data stream. So the
question arises, what compression (if any) would you use? No compression,
no way you'll get 50 users on there. How much Data is on the link sharing
bandwidth with the VoIP traffic, or is it a dedicated link for VoIP? If
there's data, do you have QoS on it? Do you have Modem lines? Fax and
Modem lines are more susceptible to compression then voice. What kind of
jitter, and latency do you have on these T1s? How susceptible are they to
packet loss?

So, in other words, there are so many possible variables to consider to make
VoIP work that you won't get them all in a note, and consultants charge
around $5,000 per site just to evaluate it.

JT


"Walter Roberson" <(E-Mail Removed)-cnrc.gc.ca> wrote in message
news:c64ujj$4j2$(E-Mail Removed)...
> In article <(E-Mail Removed) >,
> Trilok <(E-Mail Removed)> wrote:
> : I am an IT graduate student and doing a term paper on Voip
> :implementation.
>
> : 1.How many phone & fax lines can be accomodated on the T1 line?
>
> A channelized T1 would have 24 independant timeslots, 1.544 megabits
> per second total. Each timeslot could be used to carry a telephone-
> company quality call of 8000 samples per second, 8 bits per sample.
> (The extra 8000 bits are used to carry control information.)
>
> That's for standard calls. VOIP would, though, typically use IP
> and compression techniques to reduce the data stream requirements,
> and for VOIP you wouldn't necessarily want to channelize your T1.
>
> : 2.Will the T1 line be able to handle all 50 users simultaneously?
>
> You can always adjust the VOIP lossy compression algorithm parameters
> until the data fits. You need a minimum-quality metric in order to
> make a decision about how many VOIP can be carried.
>
> : 3.How do you calculate the above(1) & (2)?
>
> (1) is by specification of T1, which you can research in IETF
> standards (or just look up on some page or other at cisco.com)
>
> (2) is the much more difficult question, as it depends upon the
> quality of your perceptual coding algorithms and upon your standards
> of intelligability at the other end. It also depends on whether the
> VOIP is truly being used to carry -voice-, or if sometimes you want
> to run fax over it, or if sometimes you want high-quality music...
>
>
> : 4.What is point-to-point T1 line: does it mean that there is a
> : dedicated line that runs between one location's Lan to the
> : other location's Lan?
>
> Yup, pretty much. It might go through some switching equipment at
> various telco's along the way, but there would be a dedicated circuit
> (and probably a dedicated timeslot) on each and every one of those
> switches)
>
> : or there is a dedicated line between one location & the internet
> : backbone (ib) & from the ib to the other location?
>
> Not for a point-to-point line. But there are variations of that
> approach such as ATM in which what one gets is dedicated virtual
> circuits. Point-to-point T1's always connect the same two locations;
> virtual circuits in some of the other technologies allow bandwidth
> guarantees to be established for the duration of a session, with
> the endpoints being determinable dynamically (provided the endpoints
> are both in the service area of the technology.)
>
> : 5.How does one go about figuring out the cost of
> :hardware,software,manpower & time to implement Voip in the company
> :XYZ?
>
> One hires a consultant who has done it before. If you try to do
> VOIP with people who are unfamiliar with the technology and haven't
> had to do similar real-time work before, chances are excellent that
> mistakes will be made, configurations will be experimented with,
> the wrong equipment will be bought, the wrong dedicated line type will
> be installed (on a multi-year contract), the internal switches won't
> be upgraded to QoS, or won't be upgraded to handle enough simultaneous
> channels... etc., etc.. So if you aren't hiring a consultant to
> determine all these prices on your behalf, then whatever figure you
> come up with yourself, you had better multiply by about 8 for hardware
> and change the implimentation time to "person-years" where you had
> "person-months" before.
> --
> Are we *there* yet??



 
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shope
Guest
Posts: n/a
 
      04-21-2004

"Joe Technician" <(E-Mail Removed)> wrote in message
news:hinhc.63945$dg7.45876@edtnps84...
> Questions questions questions.
>
> The point to point T1 for Data would just be a 1.544M data stream. So the
> question arises, what compression (if any) would you use? No compression,
> no way you'll get 50 users on there.


you need to be careful what you are talking about here. The number of users
at the site may not be the same as the number of simultaneous calls you
support offsite.

e.g. if this is a call centre, then there should be 1 external voice "line"
per agent or more (or you cant run the call centre at full load, or cant Q
waiting calls etc).

if it is a business site, then there are normally fewer lines than users -
1:4 is a ratio sometimes used for offices at work, but it really depends on
how much phone use is likely, how many calls go outside, how near the worst
case peak you want to allow for.....

Also, the scenario the OP described doesnt mention where / when calls go out
to the PSTN - given 3 sites there could be a public voice connection at each
site, just at 1 site or another combination - the arrangement will change
how much of the total offsite voice traffic needs to go down each T1 link.

How much Data is on the link sharing
> bandwidth with the VoIP traffic, or is it a dedicated link for VoIP? If
> there's data, do you have QoS on it? Do you have Modem lines? Fax and
> Modem lines are more susceptible to compression then voice. What kind of
> jitter, and latency do you have on these T1s? How susceptible are they to
> packet loss?


And what is the voice encoding? if you look at the cisco design guides for
call manager, they assume G.711 for LAN calls, and G.729 for WAN calls. The
bandwidths would work out at 80k and 14 to 28k per call respectively (both
are full duplex)

have a look at the call manager design docs at
www.cisco.com/go/srnd

they will at least give you an impression of what such a design would look
like.
>
> So, in other words, there are so many possible variables to consider to

make
> VoIP work that you won't get them all in a note, and consultants charge
> around $5,000 per site just to evaluate it.
>
> JT
>
>
> "Walter Roberson" <(E-Mail Removed)-cnrc.gc.ca> wrote in message
> news:c64ujj$4j2$(E-Mail Removed)...
> > In article <(E-Mail Removed) >,
> > Trilok <(E-Mail Removed)> wrote:
> > : I am an IT graduate student and doing a term paper on Voip
> > :implementation.
> >
> > : 1.How many phone & fax lines can be accomodated on the T1 line?
> >
> > A channelized T1 would have 24 independant timeslots, 1.544 megabits
> > per second total. Each timeslot could be used to carry a telephone-
> > company quality call of 8000 samples per second, 8 bits per sample.
> > (The extra 8000 bits are used to carry control information.)
> >
> > That's for standard calls. VOIP would, though, typically use IP
> > and compression techniques to reduce the data stream requirements,
> > and for VOIP you wouldn't necessarily want to channelize your T1.
> >
> > : 2.Will the T1 line be able to handle all 50 users simultaneously?
> >
> > You can always adjust the VOIP lossy compression algorithm parameters
> > until the data fits. You need a minimum-quality metric in order to
> > make a decision about how many VOIP can be carried.
> >
> > : 3.How do you calculate the above(1) & (2)?
> >
> > (1) is by specification of T1, which you can research in IETF
> > standards (or just look up on some page or other at cisco.com)
> >
> > (2) is the much more difficult question, as it depends upon the
> > quality of your perceptual coding algorithms and upon your standards
> > of intelligability at the other end. It also depends on whether the
> > VOIP is truly being used to carry -voice-, or if sometimes you want
> > to run fax over it, or if sometimes you want high-quality music...
> >
> >
> > : 4.What is point-to-point T1 line: does it mean that there is a
> > : dedicated line that runs between one location's Lan to the
> > : other location's Lan?
> >
> > Yup, pretty much. It might go through some switching equipment at
> > various telco's along the way, but there would be a dedicated circuit
> > (and probably a dedicated timeslot) on each and every one of those
> > switches)
> >
> > : or there is a dedicated line between one location & the internet
> > : backbone (ib) & from the ib to the other location?
> >
> > Not for a point-to-point line. But there are variations of that
> > approach such as ATM in which what one gets is dedicated virtual
> > circuits. Point-to-point T1's always connect the same two locations;
> > virtual circuits in some of the other technologies allow bandwidth
> > guarantees to be established for the duration of a session, with
> > the endpoints being determinable dynamically (provided the endpoints
> > are both in the service area of the technology.)
> >
> > : 5.How does one go about figuring out the cost of
> > :hardware,software,manpower & time to implement Voip in the company
> > :XYZ?
> >
> > One hires a consultant who has done it before. If you try to do
> > VOIP with people who are unfamiliar with the technology and haven't
> > had to do similar real-time work before, chances are excellent that
> > mistakes will be made, configurations will be experimented with,
> > the wrong equipment will be bought, the wrong dedicated line type will
> > be installed (on a multi-year contract), the internal switches won't
> > be upgraded to QoS, or won't be upgraded to handle enough simultaneous
> > channels... etc., etc.. So if you aren't hiring a consultant to
> > determine all these prices on your behalf, then whatever figure you
> > come up with yourself, you had better multiply by about 8 for hardware
> > and change the implimentation time to "person-years" where you had
> > "person-months" before.
> > --
> > Are we *there* yet??

--
Regards

Stephen Hope - return address needs fewer xxs


 
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Joe Technician
Guest
Posts: n/a
 
      04-22-2004
a) I was going by his statement of 50 simultaneous users.
b) VoIP sucks just a little too much for me to even consider putting Agent
phones on VoIP sets. You have some guy doing a 50M download on a service
with no QoS and everything goes down the tubes.

JT

"shope" <(E-Mail Removed)> wrote in message
news:9Gyhc.84$B21.11@newsfe1-win...
>
> "Joe Technician" <(E-Mail Removed)> wrote in message
> news:hinhc.63945$dg7.45876@edtnps84...
> > Questions questions questions.
> >
> > The point to point T1 for Data would just be a 1.544M data stream. So

the
> > question arises, what compression (if any) would you use? No

compression,
> > no way you'll get 50 users on there.

>
> you need to be careful what you are talking about here. The number of

users
> at the site may not be the same as the number of simultaneous calls you
> support offsite.
>
> e.g. if this is a call centre, then there should be 1 external voice

"line"
> per agent or more (or you cant run the call centre at full load, or cant Q
> waiting calls etc).
>
> if it is a business site, then there are normally fewer lines than users -
> 1:4 is a ratio sometimes used for offices at work, but it really depends

on
> how much phone use is likely, how many calls go outside, how near the

worst
> case peak you want to allow for.....
>
> Also, the scenario the OP described doesnt mention where / when calls go

out
> to the PSTN - given 3 sites there could be a public voice connection at

each
> site, just at 1 site or another combination - the arrangement will change
> how much of the total offsite voice traffic needs to go down each T1 link.
>
> How much Data is on the link sharing
> > bandwidth with the VoIP traffic, or is it a dedicated link for VoIP? If
> > there's data, do you have QoS on it? Do you have Modem lines? Fax and
> > Modem lines are more susceptible to compression then voice. What kind

of
> > jitter, and latency do you have on these T1s? How susceptible are they

to
> > packet loss?

>
> And what is the voice encoding? if you look at the cisco design guides for
> call manager, they assume G.711 for LAN calls, and G.729 for WAN calls.

The
> bandwidths would work out at 80k and 14 to 28k per call respectively (both
> are full duplex)
>
> have a look at the call manager design docs at
> www.cisco.com/go/srnd
>
> they will at least give you an impression of what such a design would look
> like.
> >
> > So, in other words, there are so many possible variables to consider to

> make
> > VoIP work that you won't get them all in a note, and consultants charge
> > around $5,000 per site just to evaluate it.
> >
> > JT
> >
> >
> > "Walter Roberson" <(E-Mail Removed)-cnrc.gc.ca> wrote in message
> > news:c64ujj$4j2$(E-Mail Removed)...
> > > In article <(E-Mail Removed) >,
> > > Trilok <(E-Mail Removed)> wrote:
> > > : I am an IT graduate student and doing a term paper on Voip
> > > :implementation.
> > >
> > > : 1.How many phone & fax lines can be accomodated on the T1 line?
> > >
> > > A channelized T1 would have 24 independant timeslots, 1.544 megabits
> > > per second total. Each timeslot could be used to carry a telephone-
> > > company quality call of 8000 samples per second, 8 bits per sample.
> > > (The extra 8000 bits are used to carry control information.)
> > >
> > > That's for standard calls. VOIP would, though, typically use IP
> > > and compression techniques to reduce the data stream requirements,
> > > and for VOIP you wouldn't necessarily want to channelize your T1.
> > >
> > > : 2.Will the T1 line be able to handle all 50 users simultaneously?
> > >
> > > You can always adjust the VOIP lossy compression algorithm parameters
> > > until the data fits. You need a minimum-quality metric in order to
> > > make a decision about how many VOIP can be carried.
> > >
> > > : 3.How do you calculate the above(1) & (2)?
> > >
> > > (1) is by specification of T1, which you can research in IETF
> > > standards (or just look up on some page or other at cisco.com)
> > >
> > > (2) is the much more difficult question, as it depends upon the
> > > quality of your perceptual coding algorithms and upon your standards
> > > of intelligability at the other end. It also depends on whether the
> > > VOIP is truly being used to carry -voice-, or if sometimes you want
> > > to run fax over it, or if sometimes you want high-quality music...
> > >
> > >
> > > : 4.What is point-to-point T1 line: does it mean that there is a
> > > : dedicated line that runs between one location's Lan to the
> > > : other location's Lan?
> > >
> > > Yup, pretty much. It might go through some switching equipment at
> > > various telco's along the way, but there would be a dedicated circuit
> > > (and probably a dedicated timeslot) on each and every one of those
> > > switches)
> > >
> > > : or there is a dedicated line between one location & the internet
> > > : backbone (ib) & from the ib to the other location?
> > >
> > > Not for a point-to-point line. But there are variations of that
> > > approach such as ATM in which what one gets is dedicated virtual
> > > circuits. Point-to-point T1's always connect the same two locations;
> > > virtual circuits in some of the other technologies allow bandwidth
> > > guarantees to be established for the duration of a session, with
> > > the endpoints being determinable dynamically (provided the endpoints
> > > are both in the service area of the technology.)
> > >
> > > : 5.How does one go about figuring out the cost of
> > > :hardware,software,manpower & time to implement Voip in the company
> > > :XYZ?
> > >
> > > One hires a consultant who has done it before. If you try to do
> > > VOIP with people who are unfamiliar with the technology and haven't
> > > had to do similar real-time work before, chances are excellent that
> > > mistakes will be made, configurations will be experimented with,
> > > the wrong equipment will be bought, the wrong dedicated line type will
> > > be installed (on a multi-year contract), the internal switches won't
> > > be upgraded to QoS, or won't be upgraded to handle enough simultaneous
> > > channels... etc., etc.. So if you aren't hiring a consultant to
> > > determine all these prices on your behalf, then whatever figure you
> > > come up with yourself, you had better multiply by about 8 for hardware
> > > and change the implimentation time to "person-years" where you had
> > > "person-months" before.
> > > --
> > > Are we *there* yet??

> --
> Regards
>
> Stephen Hope - return address needs fewer xxs
>
>



 
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Mitel Lurker
Guest
Posts: n/a
 
      04-22-2004
In article <I2Fhc.2317$mP2.2000@edtnps89> "Joe Technician"
<(E-Mail Removed)> writes:

>a) I was going by his statement of 50 simultaneous users.
>b) VoIP sucks just a little too much for me to even consider putting Agent
>phones on VoIP sets. You have some guy doing a 50M download on a service
>with no QoS and everything goes down the tubes.


anyone who attempts to do VOIP on a data network needs to create at least
two Vlans; one exclusively for voice and all other(s) for data. Without
some form of QOS on a network w/mixed traffic, you're an idiot and doomed
for failure.

 
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Joe Technician
Guest
Posts: n/a
 
      04-22-2004
The problem with your scenario is the customers get suckered into the
propaganda from Nortel, Cisco, Avaya, Siemens, Mitel, etc. which says that
VoIP will work on their LAN as it sits. Then when they're told they need a
separate V-LAN for voice, they flip claiming that's not what the vendors
told them. Of course, V-LAN isn't the only way to get QoS.

JT


"Mitel Lurker" <wdg@[206.180.145.133]> wrote in message
news:(E-Mail Removed)...
> In article <I2Fhc.2317$mP2.2000@edtnps89> "Joe Technician"
> <(E-Mail Removed)> writes:
>
> >a) I was going by his statement of 50 simultaneous users.
> >b) VoIP sucks just a little too much for me to even consider putting

Agent
> >phones on VoIP sets. You have some guy doing a 50M download on a service
> >with no QoS and everything goes down the tubes.

>
> anyone who attempts to do VOIP on a data network needs to create at least
> two Vlans; one exclusively for voice and all other(s) for data. Without
> some form of QOS on a network w/mixed traffic, you're an idiot and doomed
> for failure.
>



 
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Joe Matuscak
Guest
Posts: n/a
 
      04-22-2004
In article <oIFhc.2447$mP2.1105@edtnps89>, http://www.velocityreviews.com/forums/(E-Mail Removed) says...
> The problem with your scenario is the customers get suckered into the
> propaganda from Nortel, Cisco, Avaya, Siemens, Mitel, etc. which says that
> VoIP will work on their LAN as it sits. Then when they're told they need a
> separate V-LAN for voice, they flip claiming that's not what the vendors
> told them. Of course, V-LAN isn't the only way to get QoS.


I dont know about the rest of them, but Cisco certainly mentions the
idea of using a VLAN for the phones if for no other reason than to get a
seperate IP range.

--
Joe Matuscak
Rohrer Corporation
717 Seville Road
Wadsworth, OH 44281

 
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shope
Guest
Posts: n/a
 
      04-22-2004

"Mitel Lurker" <wdg@[206.180.145.133]> wrote in message
news:(E-Mail Removed)...
> In article <I2Fhc.2317$mP2.2000@edtnps89> "Joe Technician"
> <(E-Mail Removed)> writes:
>
> >a) I was going by his statement of 50 simultaneous users.
> >b) VoIP sucks just a little too much for me to even consider putting

Agent
> >phones on VoIP sets. You have some guy doing a 50M download on a service
> >with no QoS and everything goes down the tubes.


what i see here in the UK is that a lot of call centres are moving to IP
telephony (and dont seem to hit any more problems than the TDM kind) - but
your area may be different
>
> anyone who attempts to do VOIP on a data network needs to create at least
> two Vlans; one exclusively for voice and all other(s) for data. Without
> some form of QOS on a network w/mixed traffic, you're an idiot and doomed
> for failure.


right - but the real fundamental is QoS (and call admission control over any
lower bandwidth link so the voice stuff cant contend with itself).

the vlan stuff is best practice on a LAN, and it does make traffic
separation and various security things easier, since you can filter by
subnet if need be. But voice traffic is usually possible to separate out by
protocol if need be. (you more or less have to use integrated voice and data
LANs with softphones).
--
Regards

Stephen Hope - return address needs fewer xxs


 
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Mitel Lurker
Guest
Posts: n/a
 
      04-23-2004
In article <oIFhc.2447$mP2.1105@edtnps89> "Joe Technician"
<(E-Mail Removed)> writes:

> The problem with your scenario is the customers get suckered into the
>propaganda from Nortel, Cisco, Avaya, Siemens, Mitel, etc. which says that
>VoIP will work on their LAN as it sits. Then when they're told they need a
>separate V-LAN for voice, they flip claiming that's not what the vendors
>told them. Of course, V-LAN isn't the only way to get QoS.


VOIP -will- work on the LAN as it sits, it just won't work very well. I
thought Mitel made that pretty clear in their PowerPoint presentation.

What really annoys me is the Cisco presentation mentions nothing about the
fact that their whole she-bang runs on a collection of Microsoft servers
running a Cisco-mutated load of Win2K server and SQL. Mother of God! The
mere concept of your mission-critical phone system being dependent upon
MS-anything is a haunting thought. Cisco also fails to mention that you'll
need to replace or upgrade or add memory to all of your existing routers,
something which *is not* necessary with Mitel's 3300 ICP system.

Ask Cisco how they handle voice network security and they'll point you
towards *ANOTHER* freakin' MS box acting as a firewall. So you say you
need voice mail??? Well okay, but that means ***ANOTHER*** (!!!!) Win2K
box for Unity. Voice mail is built-in on the Mitel, 20 ports & 400 hours
default, expandable to 30 ports.

Ask Cisco how many line key appearances they offer on their most robust IP
desktop instrument. Better hope you or your admins don't need more than 6,
'cause that's the limit. With Mitel's 5220IP set (14 lines all by itself)
you can go to 62 lines with the addition of a PKM48 module and all the way
to 110 lines with 2 PKM48s. While that makes somewhat of a church
organ-sized phone, if you need the line appearances or keys, Mitel can
deliver 'em.

Need to do PS/ALI to meet certain state-mandated delivery of station
location information to your 9-1-1 regional PSAP? It's already built into
and enabled in the Mitel 3300 base load at no extra charge. Guess what
you'll need to do it with the Cisco? That's right, another MS-based
server.

Are you beginning to see a pattern developing here?


 
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