Voip implementation

Discussion in 'VOIP' started by Trilok, Apr 21, 2004.

  1. Trilok

    Trilok Guest

    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
     
    Trilok, Apr 21, 2004
    #1
    1. Advertising

  2. In article <>,
    Trilok <> 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??
     
    Walter Roberson, Apr 21, 2004
    #2
    1. Advertising

  3. 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" <-cnrc.gc.ca> wrote in message
    news:c64ujj$4j2$...
    > In article <>,
    > Trilok <> 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??
     
    Joe Technician, Apr 21, 2004
    #3
  4. Trilok

    shope Guest

    "Joe Technician" <> 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" <-cnrc.gc.ca> wrote in message
    > news:c64ujj$4j2$...
    > > In article <>,
    > > Trilok <> 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
     
    shope, Apr 21, 2004
    #4
  5. 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" <> wrote in message
    news:9Gyhc.84$B21.11@newsfe1-win...
    >
    > "Joe Technician" <> wrote in message
    > news:hinhc.63945$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" <-cnrc.gc.ca> wrote in message
    > > news:c64ujj$4j2$...
    > > > In article <>,
    > > > Trilok <> 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
    >
    >
     
    Joe Technician, Apr 22, 2004
    #5
  6. Trilok

    Mitel Lurker Guest

    In article <I2Fhc.2317$mP2.2000@edtnps89> "Joe Technician"
    <> 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.
     
    Mitel Lurker, Apr 22, 2004
    #6
  7. 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:...
    > In article <I2Fhc.2317$mP2.2000@edtnps89> "Joe Technician"
    > <> 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.
    >
     
    Joe Technician, Apr 22, 2004
    #7
  8. Trilok

    Joe Matuscak Guest

    In article <oIFhc.2447$mP2.1105@edtnps89>, 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
     
    Joe Matuscak, Apr 22, 2004
    #8
  9. Trilok

    shope Guest

    "Mitel Lurker" <wdg@[206.180.145.133]> wrote in message
    news:...
    > In article <I2Fhc.2317$mP2.2000@edtnps89> "Joe Technician"
    > <> 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
     
    shope, Apr 22, 2004
    #9
  10. Trilok

    Mitel Lurker Guest

    In article <oIFhc.2447$mP2.1105@edtnps89> "Joe Technician"
    <> 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?
     
    Mitel Lurker, Apr 23, 2004
    #10
  11. In article <>,
    Mitel Lurker <wdg@[206.180.145.133]> wrote:
    :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.

    {Sputter.} We were doing more than 6 [pure-digital] line appearances
    20 years ago when I was working on the Mitel SX-2000. The
    figure that sticks in my mind is that my simulation software (for
    demonstrating the console at trade shows) had 8 line appearances
    and that at the time our hardcoded limit was 20 [but that made for
    a crowded display, with all the bar graphs showing the size of the
    call queues.)

    If I recall correctly (don't count on it!), we were running the whole show
    off of a pair of 68020's @ 20 MHz each, per line rack, plus some DSPs.
    (I covetted those 68020's -- my 68000 machine at home only ran at 8 MHz,
    and 20 MHz was so much nicer...) Our development systems were Vaxen,
    780's as I recall. The SX2000 boot images were stored on bubble memory...

    But my NDA probably in theory still prohibits me from talking about
    which language we programmed it all in ;-)
    --
    "[...] it's all part of one's right to be publicly stupid." -- Dave Smey
     
    Walter Roberson, Apr 23, 2004
    #11
  12. Trilok

    Mitel Lurker Guest

    In article <c6a1fq$aea$>
    -cnrc.gc.ca (Walter Roberson) writes:

    >{Sputter.} We were doing more than 6 [pure-digital] line appearances
    >20 years ago when I was working on the Mitel SX-2000.


    Errrr.... that would be E-stream, MS-2001 as memory serves me. It was a
    15-line set using digital signalling and control superimposed on the
    analog voice channel. See? You're not the only old fart around here. If
    you really are that old then you must surely remember the "Lew Barnhouse
    College of SX2000 Knowledge" - 5400 Broken Sound Blvd, Boca Raton, FL.
    Still got my original cert with his signature on it.... or was it Mike and
    Terry's Lawnmower Service... I forget >>smile<<

    >If I recall correctly (don't count on it!), we were running the whole show
    >off of a pair of 68020's @ 20 MHz each, per line rack, plus some DSPs.


    68020 @ 20 Mhz is correct.

    >But my NDA probably in theory still prohibits me from talking about
    >which language we programmed it all in ;-)


    Wouldn't have been ADA, would it?

    And while here, I need to correct my previous post. Cisco actually does
    have some side modules similar in concept to the PKM that can extend their
    6-line high end IP set to accomodate addt'l line keys. Unfortunately their
    basic 6-line set is near $400 compared to the 14-line Mitel 5220 at barely
    over $200.
     
    Mitel Lurker, Apr 24, 2004
    #12
  13. In article <>,
    Mitel Lurker <wdg@[206.180.145.133]> wrote:
    :In article <c6a1fq$aea$>
    :-cnrc.gc.ca (Walter Roberson) writes:

    :>{Sputter.} We were doing more than 6 [pure-digital] line appearances
    :>20 years ago when I was working on the Mitel SX-2000.

    :Errrr.... that would be E-stream, MS-2001 as memory serves me. It was a
    :15-line set using digital signalling and control superimposed on the
    :analog voice channel.

    I don't remember the official designation. I'm pretty sure it wasn't
    MS-2001. We called it "The SX2000 Console". I believe you are referring
    to the -next- major product that went through, which did some hybrid
    SS7 signalling as a transition-level product.


    :See? You're not the only old fart around here. If
    :you really are that old then you must surely remember the "Lew Barnhouse
    :College of SX2000 Knowledge" - 5400 Broken Sound Blvd, Boca Raton, FL.
    :Still got my original cert with his signature on it.... or was it Mike and
    :Terry's Lawnmower Service... I forget >>smile<<

    I predate anyone in the USA knowing anything about the SX2000 ;-)

    I was an co-op/student programmer at Mitel in the year leading up to the
    SX2000 launch. As I recall, my official contribution of record was
    the debugging of the "Call Forward" protocols, and the design and
    implimentation of "Camp on - Busy" and "Camp on - No Answer". But I also
    ended up building a bunch of the development tools, taking on noticable
    portions of the systems administration, and doing a *lot* of integration
    testing and debugging -- so my fingers ended up in the pies of nearly
    everything -except- the programming of the digital line cards themselves,
    the development of the bubble-memory subsystem. I also did was not involved
    in the development of the compilers themselves, but because the nightly
    builds somehow ended up in front of me, I did end up doing some
    work with the compiler people (when the compiles failed, or the image
    didn't work, I'd be debugging the switch code or debugging the compilers...)


    :>But my NDA probably in theory still prohibits me from talking about
    :>which language we programmed it all in ;-)

    :Wouldn't have been ADA, would it?

    ADA was still in the discussion phases in that timeframe.

    Nah, my NDA probably expired about 15 years ago -- I'm just embarrased
    to name the development language ;-)


    Ah, those were the days... We had some good people, and I wonder what
    happened to some of them. I learned a lot. Even got my heart broken
    for the first time ;-)
    --
    The Knights Of The Lambda Calculus aren't dead --this is their normal form!
     
    Walter Roberson, Apr 24, 2004
    #13
  14. Trilok

    Mitel Lurker Guest

    In article <c6cmup$360$>
    -cnrc.gc.ca (Walter Roberson) writes:

    >I don't remember the official designation. I'm pretty sure it wasn't
    >MS-2001.


    There's an old TSB still on Mitel Online, doc # MD4209-NA-01 written Dec
    '98 which states in part: <quoted>

    1.1 MS2001 (E10)
    This release introduced the following:
    Hardware
    " Supersetâ„¢3, Supersetâ„¢ 4 & Supersetâ„¢7 introduced
    " Dataset 1 & 2 introduced
    " Redundant File System developed
    " Almost every control card required major hardware & firmware rev
    changes from C-stream including the COV cards.
    " Double ribbon cable on the winchester disk drives
    " Greater than 1 Peripheral Pair
    Maintenance
    " "restrict" Command to load balance Per Pairs (Rotary sets & DID
    Trunks)
    Features
    " Advanced ARS
    " Advanced Data
    " SMDR

    <end quote>

    The reference here of course being to officially released, shipping
    hardware and program code...

    >I was an co-op/student programmer at Mitel in the year leading up to the
    >SX2000 launch.


    That would had to have been 1982 then. The SX2000 was officially released
    to production in 1983 with software level C17 (MS2000).

    Quoting again from that same document....

    4.1 MS2000 (C17)
    The SX-2000 finally exited field trials with C17 software. There was
    limited functionality in this release. There was no redundant file system
    on this single cabinet SG.
    " Hardware
    " SG - Single Cabinet, Non-Redundant
    " Features
    " Limited Functionality (Basic Voice)

    <end quote>

    The multiline "digital sets" (Superset 3 and 4) did not come along
    (officially) until MS2001 and software stream release E10.4 in 1984.
    (there was no D-stream). Somewhere I heard rumors of a "Superset 1" but
    I've never seen one and have been so far unable to confirm its existence.

    >Nah, my NDA probably expired about 15 years ago -- I'm just embarrased
    >to name the development language ;-)


    Ahah, so you know COBOL!!! You know of course this means you will have to
    be cryogenically frozen until the year 2999 so you can be thawed out at
    that time to help mankind deal with transitioning their computers of the
    era to Y3K ;-)

    >Ah, those were the days... We had some good people, and I wonder what
    >happened to some of them. I learned a lot. Even got my heart broken
    >for the first time ;-)


    MaryJo? (drool....)
     
    Mitel Lurker, Apr 24, 2004
    #14
  15. In article <>,
    Mitel Lurker <wdg@[206.180.145.133]> wrote:
    :The multiline "digital sets" (Superset 3 and 4) did not come along
    :(officially) until MS2001 and software stream release E10.4 in 1984.
    :(there was no D-stream). Somewhere I heard rumors of a "Superset 1" but
    :I've never seen one and have been so far unable to confirm its existence.

    Now that name, "Superset 1", sounds familiar in the timeframe I was
    there. Somewhere in the back of my mind, some neurons are firing
    suggesting to me that the Superset 1 was released first in the UK.

    I am *sure* the SX-2000 console, whatever it's official model number,
    was multi-line and pure digital -- I worked with it a bit on
    some throw-away code for the launch demo.


    :Ahah, so you know COBOL!!!

    PERFORM MYLIFE VARYING DENY_KNOWING_COBOL TO TRUE.


    :>Ah, those were the days... We had some good people, and I wonder what
    :>happened to some of them. I learned a lot. Even got my heart broken
    :>for the first time ;-)

    :MaryJo? (drool....)

    I didn't work with any "MaryJo". Sounds like a name from the Corporate
    side. No, a fair bit of the development team was not much older than I --
    if you had 6 years of experience you were entitled to be a "senior
    programmer" (3rd official ranking up, not including students); pretty
    much anyone over the age of 26 was a manager, and by 33 you would
    definitely be a senior manager, perhaps with 2 levels below you (and
    the students below that.) But as they say, some comrades are more equal
    than others, and to be a senior development manager was certainly to
    have much less real power and prestige than a more junior manager on
    the corporate side.


    > 1982


    January 1982 to mid-September 1983 for me. It was an interesting time
    to work there (the Kanata development lab). I went back to university
    after that, which was a bit tough to re-adapt to after many months of
    very long days doing real development work. I could have stayed, and
    life would have been very different if I had; but I heard the whole
    Mitel corporate culture changed a *lot* over the two years after that
    (high inflation, low sales, OPEC crisis.) By 1985 or so, some of
    the people I had worked with had left Mitel and were saying they'd
    never go back.

    If it had all happened a dozen years later, it would have been a
    typical dot-com experience -- work hard, almost cultish-ly so, with
    some great people, to get out A Product That Will Change The World...
    only to have the product not do so well, the whole culture turn sour
    compared to the early days, big layoffs, talk of bankruptcy, founders
    forced out, people burnt out, etc..


    And now that you have me going over my memories of that time, and having
    gained a bit of perspective, I think I'd say that experiences like that
    are addictive -- there's a real "rush" in feeling like you're working
    hard for a Good Cause, that the problems are tough but your team is is
    bright and conquers them, that you belong, that what you are doing
    is Important, and that you yourself are Important because you are
    giving so much of yourself to the Cause.
    --
    This signature intentionally left... Oh, darn!
     
    Walter Roberson, Apr 24, 2004
    #15
  16. Trilok

    Mitel Lurker Guest

    In article <c6e6cn$741$>
    -cnrc.gc.ca (Walter Roberson) writes:


    >I am *sure* the SX-2000 console, whatever it's official model number,
    >was multi-line and pure digital -- I worked with it a bit on
    >some throw-away code for the launch demo.


    The first SX2000 "Console" was in fact pure digital and the PBX card that
    ran it was called a "High Speed Digital Line Card" and supported a max of
    8 consoles, but only a fool ever dared to put more than 4 on one card due
    to bus congestion on the backplane, a limitation which never improved
    throughout the entire life of the SG machine. The released/shipping
    console of that era was called the "Superset 7". Believe it or not I just
    finally retired 3 of those things back in 2001 when we (finally) replaced
    5 cabinets of SG hardware (2 main controls and total of 3 PER cabinets)
    with 34 cabinets of SX2K Light (4 main controls). The SX2K Light
    architecture successfully compressed a full entire SX2000SG "Control
    Plane" (line rack) down to TWO plug in boards. The new MN3300 ICP machine
    squeezes the whole freakin switch down to "2U" in a 19" rack (700 lines
    present max cap. of the 3300)

    >:Ahah, so you know COBOL!!!


    >PERFORM MYLIFE VARYING DENY_KNOWING_COBOL TO TRUE.


    ROTFLMAO!

    >January 1982 to mid-September 1983 for me. It was an interesting time
    >to work there (the Kanata development lab).


    The Kanata lab is still there and going strong. They sold off the
    semiconductor devision to pump capital back into the PBX. Terry Matthews
    has been back at the helm for a couple years now and has breathed some new
    life into the company, hired a bunch of engineers, etc. If 9-1-1 and the
    Enron debacle hadn't happened, it would have gone public by now. In case
    you didn't know, they bought Gandalf and March Networks a while back,
    which of course became the impetus for the new MN3300 VOIP product.

    Did you happen to know a gent by the name of Claude Richard?
     
    Mitel Lurker, Apr 25, 2004
    #16
  17. Trilok

    John Guest

    Ahh you Mitel boys bring a tear to my eye. Recently I removed an SX-200 I
    maintained and replaced it with an option 11. I watched as the trash men
    hauled her away since no home was to be found for, even her parts.

    John 807
    "Mitel Lurker" <wdg@[206.180.145.133]> wrote in message
    news:...
    > In article <c6e6cn$741$>
    > -cnrc.gc.ca (Walter Roberson) writes:
    >
    >
    > >I am *sure* the SX-2000 console, whatever it's official model number,
    > >was multi-line and pure digital -- I worked with it a bit on
    > >some throw-away code for the launch demo.

    >
    > The first SX2000 "Console" was in fact pure digital and the PBX card that
    > ran it was called a "High Speed Digital Line Card" and supported a max of
    > 8 consoles, but only a fool ever dared to put more than 4 on one card due
    > to bus congestion on the backplane, a limitation which never improved
    > throughout the entire life of the SG machine. The released/shipping
    > console of that era was called the "Superset 7". Believe it or not I just
    > finally retired 3 of those things back in 2001 when we (finally) replaced
    > 5 cabinets of SG hardware (2 main controls and total of 3 PER cabinets)
    > with 34 cabinets of SX2K Light (4 main controls). The SX2K Light
    > architecture successfully compressed a full entire SX2000SG "Control
    > Plane" (line rack) down to TWO plug in boards. The new MN3300 ICP machine
    > squeezes the whole freakin switch down to "2U" in a 19" rack (700 lines
    > present max cap. of the 3300)
    >
    > >:Ahah, so you know COBOL!!!

    >
    > >PERFORM MYLIFE VARYING DENY_KNOWING_COBOL TO TRUE.

    >
    > ROTFLMAO!
    >
    > >January 1982 to mid-September 1983 for me. It was an interesting time
    > >to work there (the Kanata development lab).

    >
    > The Kanata lab is still there and going strong. They sold off the
    > semiconductor devision to pump capital back into the PBX. Terry Matthews
    > has been back at the helm for a couple years now and has breathed some new
    > life into the company, hired a bunch of engineers, etc. If 9-1-1 and the
    > Enron debacle hadn't happened, it would have gone public by now. In case
    > you didn't know, they bought Gandalf and March Networks a while back,
    > which of course became the impetus for the new MN3300 VOIP product.
    >
    > Did you happen to know a gent by the name of Claude Richard?
    >
     
    John, Apr 26, 2004
    #17
  18. Trilok

    Mitel Lurker Guest

    Assuming that was an old SX200 Analog or an analog-converted digital
    (Black box), don't apologize, you did the right thing. Not even the Mitel
    secondary market salvage shops have any use for that old iron. Oh maybe
    Mike Sandman in Chicago would have taken it. On the off chance it was a
    200ML or 200EL, Terry Cunningham at Dean Enterprises would have given you
    something for it. The 200ML/EL is a nice little system, ideal hotel/motel
    or small to medium size office.

    Where I come from an "807" is either an old power tetrode or a can of
    beer.

    In article <GYZic.27598$> "John"
    <> writes:

    >Ahh you Mitel boys bring a tear to my eye. Recently I removed an SX-200 I
    >maintained and replaced it with an option 11. I watched as the trash men
    >hauled her away since no home was to be found for, even her parts.
    >
    > John 807
     
    Mitel Lurker, Apr 28, 2004
    #18
  19. Trilok

    John Guest

    Yeah, it was an old analog sx-200 that some unauthorized snake oil salesman
    sold the customer about 4 years ago claiming it was new then disappeared.
    Through a friend of a friend I ended up servicing the account after
    informing the customer the switch was '80's vintage and giving him the
    option to upgrade then. Since he had just bought the switch we limped it
    along until the batteries on the gen. 217 died and he didn't want to replace
    and reprogram.

    "807" old Bell Atlantic I.D. number (take away your name and give you
    a number) Name that song?


    John 807
    "Mitel Lurker" <wdg@[206.180.145.133]> wrote in message
    news:...
    > Assuming that was an old SX200 Analog or an analog-converted digital
    > (Black box), don't apologize, you did the right thing. Not even the Mitel
    > secondary market salvage shops have any use for that old iron. Oh maybe
    > Mike Sandman in Chicago would have taken it. On the off chance it was a
    > 200ML or 200EL, Terry Cunningham at Dean Enterprises would have given you
    > something for it. The 200ML/EL is a nice little system, ideal hotel/motel
    > or small to medium size office.
    >
    > Where I come from an "807" is either an old power tetrode or a can of
    > beer.
    >
    > In article <GYZic.27598$> "John"
    > <> writes:
    >
    > >Ahh you Mitel boys bring a tear to my eye. Recently I removed an SX-200 I
    > >maintained and replaced it with an option 11. I watched as the trash men
    > >hauled her away since no home was to be found for, even her parts.
    > >
    > > John 807

    >
     
    John, Apr 28, 2004
    #19
  20. Trilok

    Mitel Lurker Guest

    Snake oil salesmen gotta eat too. 5 years ago I sold and installed a
    4-Line 1A2 system (WECO 551 shoebox) + Melco dial intercom and a
    half-dozen refurb'd 2565 HCK's to a boat shop in Bettendorf, Iowa. The guy
    needed phones & intercom *REAL BAD* but had no money. I showed him how to
    change lamps and gave him a spare interruptor and some extra 400D cards.
    Got an xmas card from him last year, and said phones are working fine. You
    cannot kill 1A2.

    In article <jcTjc.27776$> "John"
    <> writes:

    >Yeah, it was an old analog sx-200 that some unauthorized snake oil salesman
    >sold the customer about 4 years ago claiming it was new then disappeared.
    >Through a friend of a friend I ended up servicing the account after
    >informing the customer the switch was '80's vintage and giving him the
    >option to upgrade then. Since he had just bought the switch we limped it
    >along until the batteries on the gen. 217 died and he didn't want to replace
    >and reprogram.
    >
    > "807" old Bell Atlantic I.D. number (take away your name and give you
    >a number) Name that song?
    >
    >
    > John 807
    >"Mitel Lurker" <wdg@[206.180.145.133]> wrote in message
    >news:...
    >> Assuming that was an old SX200 Analog or an analog-converted digital
    >> (Black box), don't apologize, you did the right thing. Not even the Mitel
    >> secondary market salvage shops have any use for that old iron. Oh maybe
    >> Mike Sandman in Chicago would have taken it. On the off chance it was a
    >> 200ML or 200EL, Terry Cunningham at Dean Enterprises would have given you
    >> something for it. The 200ML/EL is a nice little system, ideal hotel/motel
    >> or small to medium size office.
    >>
    >> Where I come from an "807" is either an old power tetrode or a can of
    >> beer.
    >>
    >> In article <GYZic.27598$> "John"
    >> <> writes:
    >>
    >> >Ahh you Mitel boys bring a tear to my eye. Recently I removed an SX-200 I
    >> >maintained and replaced it with an option 11. I watched as the trash men
    >> >hauled her away since no home was to be found for, even her parts.
    >> >
    >> > John 807

    >>

    >
     
    Mitel Lurker, Apr 29, 2004
    #20
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