newbie questions - Asterisk & VOIP

Discussion in 'UK VOIP' started by Geoff Winkless, Jun 1, 2005.

  1. Hi

    Firstly, some background...

    I have a certain amount of experience in traditional telecoms (I used to
    administer an Index PBX in my last job) and TCP/IP networking, plus 10
    years commercial experience in Linux and IT in general.

    My wife has her own business which now has four people working in one
    office. They currently have two external voice lines with DECT handsets
    (one for each line on each desk, so 8 phones in all, bleagh), which was
    ok when they first started out (it was just two of them in partnership)
    but now they have more people it's becoming unworkable - they need more
    lines and having more DECT phones won't cut it.

    I looked into an asterisk solution a while back but the cost of getting
    the hardware was prohibitive for a startup.

    However now they're looking at buying more physical lines it seems to me
    that VOIP might be a better solution: to get 5 analogue lines (4 voice,
    1 fax) is going to cost a pretty packet and will still require some
    hardware to interface to a PBX.

    So the questions.

    In your experience will an NTL cable internet connection (2Mb, I think)
    be sufficient to run 5 VOIP lines, given concurrent normal office use
    (ie email, basic surfing etc) or would a dedicated second IP connection
    be required?

    How easy is it to interface asterisk to VOIP supplied lines from the
    various suppliers (skype, sipgate etc)?

    Is it worth the extra to use VOIP/Wifi phones or would you get adaptors
    and use DECT? Will the DECT solution restrict usability (things like
    missing callerid, exchange features etc)? Given that adaptors are 50
    quid or so and wifi phones about 200, that's a fairly large price
    difference...

    Is it possible to link a voip phone to asterisk over the internet - so
    if I take a wifi-VOIP phone home and turn it on there (where I also have
    a wifi network) is it possible to link to the office asterisk install
    and transparently work from home? I guess I could set up a tunnel using
    my server at home but is there an out-of-the-box solution to do this?

    Lots of questions, I know, sorry :(

    Thanks for any advice

    Geoff
     
    Geoff Winkless, Jun 1, 2005
    #1
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  2. Geoff Winkless wrote:
    || Hi
    ||
    <snip>
    ||
    || In your experience will an NTL cable internet connection (2Mb, I
    || think) be sufficient to run 5 VOIP lines, given concurrent normal
    || office use (ie email, basic surfing etc) or would a dedicated second
    || IP connection be required?
    ||

    You will have to be mindfull of the upload speed available & that a high
    quality codec will use 80k per call (80?)

    || How easy is it to interface asterisk to VOIP supplied lines from the
    || various suppliers (skype, sipgate etc)?
    ||

    Skype doesn't have an Asterisk client yet, though someone is working on one,
    somewhere.
    Sipgate's a piece of cake, 1899 worked fine for me. I'm struggling with
    18866, though many others aren't

    || Is it worth the extra to use VOIP/Wifi phones or would you get
    || adaptors and use DECT? Will the DECT solution restrict usability
    || (things like missing callerid, exchange features etc)? Given that
    || adaptors are 50 quid or so and wifi phones about 200, that's a
    || fairly large price difference...
    ||

    Wouldn't bother with WiFi phone - Buy an adapter & plug in your Dect
    phone(s).

    Caller ID will still work.

    Sipura make some excellent ATAs.

    || Is it possible to link a voip phone to asterisk over the internet -

    Apparently so. I have not successfully managed to set this up.
    ||
    || Lots of questions, I know, sorry :(

    That's what the group's for.

    An excellent resource is http://voxilla.com they have some very helpful
    forum moderators, who also know what they're talking about.

    There's a forum for Adapters and another for Asterisk (plus others)
     
    The Cable Guy, Jun 1, 2005
    #2
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  3. Geoff Winkless

    Ian Guest

    Depends on the upstream speed. also depends on the call usage each call will
    use approx 80kbs for G711 and 20kbs for G729, Its a case realy of suck it
    and see and if you get prolems get a second line in.
    Well Skype is impossible at the moment, Sipgate is as it stands not suitable
    for business use. Reasons include reliablity, no support for IAX and DTMF
    issues
    Look to the business suppliers if you want business lines
    Dect sets will have all the features you need CLI etc
    Depends on how many dect sets you want if its more than a couple get a
    TDM400 which has up to 4 fxs ports.
    Yes, But you get issues with NAT traversal.

    for more info have a look at www.voip-info.org

    Ian
     
    Ian, Jun 1, 2005
    #3
  4. Geoff Winkless

    BigLarry Guest

    In your experience will an NTL cable internet connection (2Mb, I
    || think) be sufficient to run 5 VOIP lines, given concurrent normal
    || office use (ie email, basic surfing etc) or would a dedicated second

    || IP connection be required

    In my experience (with a 3MB NTL line) it isnt even reliable enough for
    1 line, at least for a business situation.

    The upload just isn't enough and without doubt you will at some stage
    get situations where people just cannot hear you or at best say "its
    breaking up I cant hear you".

    I have learnt this from experience.

    Take into consideration that someone might also be sending a large
    email attachment and this WILL seriously affect your phone quality.

    I'm told that moving to a 20:1 dedicated adsl contended line will allow
    up to 10 people without issues to hold concurrent calls - perhaps
    someone can confirm that.

    Hope this helps.
     
    BigLarry, Jun 4, 2005
    #4
  5. if you use the higher quality Codecs that use 10 kbytes/s per call
    then three of them would max out the upstream of a BT 256k upstream
    ADSL line.

    I don't know if the trunking facility of Asterisk reduces overheads
    enough to help in this regard.

    Phil
     
    Phil Thompson, Jun 5, 2005
    #5
  6. Geoff Winkless

    Tony Hoyle Guest

    Not to the extent of supporting 10 simultaneous calls.

    If the line is shared it needs QoS setup as well, otherwise other
    activity on the line will increase your jitter too much. This of
    course requires a router that supports it.

    TBH for a business get SDSL or Leased Line. ADSL is for home users and
    low usage.. not for something as critical as VOIP. I use it but the
    max I'd ever use is two calls and that's rarely.

    Tony
     
    Tony Hoyle, Jun 5, 2005
    #6
  7. Geoff Winkless

    Tony Hoyle Guest

    Just read that bit.

    Get an ISDN PRI and something like an TE110P, which will scale to 32
    channels. You can still use cheap asterisk Hardware and have the same
    flexibility.

    Tony
     
    Tony Hoyle, Jun 5, 2005
    #7
  8. 10kB? Surely full uncompressed G711 is 64Kbps, or 8kB.

    uh huh... http://www.voip-info.org/wiki-Bandwidth+consumption suggests
    that 64kbps stream translates to a bandwidth usage of around 87kbps,
    which is ~11kB, I suppose. However, that's full stream uncompressed,
    you'd expect to get better than that using compression.
    Given that I wouldn't be using it (unless the VOIP provider supports
    this - do any of them?) it wouldn't help in this particular instance.

    G
     
    Geoff Winkless, Jun 6, 2005
    #8
  9. Thanks, that pretty much scuppers that.

    Geoff
     
    Geoff Winkless, Jun 6, 2005
    #9
  10. that's the Vo bit, now you need to add the IP. Lots of little packets,
    lots of overheads. Not very efficient really. My 10 kbytes/s was read
    off Netmeter while I made a call, BTBBV reports similar, I didn't just
    make it up to depress you :)
    what does the compression ? Why not use GSM codec instead.
    http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2

    yes there are VoIP providers supporting Asterisk connections for
    trunking rather than just individual clients registering direct to
    their server.
    http://www.voiptalk.org/products/VoIPtalk+Services/iaxtalk (arbitrary
    example)

    Phil
     
    Phil Thompson, Jun 6, 2005
    #10
  11. Sorry, my misuse of the terms. When I said "using compression" I meant
    "using a codec which uses compresses".
    Thanks, I'll bear that in mind!

    Geoff
     
    Geoff Winkless, Jun 6, 2005
    #11
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