Asterisk@home, conferences, codecs and bandwidth

Discussion in 'UK VOIP' started by linker3000, May 21, 2006.

  1. linker3000

    linker3000 Guest

    Hi guys,

    I posted this in the A@H forums but hope I might catch a few more
    helpful people here too...

    I'd appreciate some real-world experience if possible:

    If we roll out our * trial, we will initially have about 30 IP phones
    (no analogues), one per branch office, and I was wondering about
    conference/bandwidth issues...

    The server is a dual core P4-2.6GHz with 1GiB RAM so I guess we have
    enough steam to start with?! At the moment the server is conncted to a
    2Mbit ADSL line with an upstream speed of 288Kb. I will probably upgrade
    this to 8Mbit/448Kb service.

    Here's the main questions:

    How's the system likely to cope with all 30 sites in a conference over
    the current/upgraded ADSL line?

    I understand that the choice/mix of codecs used when conferencing can
    have a big impact on the results - our current phone of choice supports
    the usual a-law and u-law stuff and others will also do ilbc and gsm -
    obviously, the latter two use less bandwidth but what is likely to
    happen if we mix them with phones on a-law/u-law or should we just stick
    to one set of common codecs - which would be the bigger bandwidth
    a/u-law ones?

    I have seen some dev threads + others on silence suppression and keeping
    it OFF in order to provide the proper timing for RTP streams - does this
    still apply with A@H 2.8?

    How does the ADSL bandwidth relate to max number of simultaneous
    connections? I can appreciate a simple calculation of (available
    bandwidth / codec bandwidth requirement), but guess in the real world
    it's not as simple as that when considering things such as simultaneous
    use of channels not actually transmitting/receiving 100% of the time,
    but how aboout the streams associated with conferences and also the
    'overhead' of transmitting all the silence?

    I have had a good look around the main Asterisk and Asterisk@Home sites
    and have found some generalised info on all of this, but not a lot of
    real world input.

    Any insight appreciated.

    Thanks
    linker3000, May 21, 2006
    #1
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  2. linker3000

    stephen Guest

    "linker3000" <> wrote in message
    news:44708a9c$0$8335$...
    > Hi guys,
    >
    > I posted this in the A@H forums but hope I might catch a few more
    > helpful people here too...
    >
    > I'd appreciate some real-world experience if possible:
    >
    > If we roll out our * trial, we will initially have about 30 IP phones
    > (no analogues), one per branch office, and I was wondering about
    > conference/bandwidth issues...
    >
    > The server is a dual core P4-2.6GHz with 1GiB RAM so I guess we have
    > enough steam to start with?! At the moment the server is conncted to a
    > 2Mbit ADSL line with an upstream speed of 288Kb. I will probably upgrade
    > this to 8Mbit/448Kb service.


    i dont know the asterisk specifics - but your link is going to limit the no.
    of simultaneous calls.

    depending on the codec you use, you are going to need 80 Kbps or more for
    G.711, or 24 Kbps for G.729.

    you cant have header compression on the link (since the ISP routers wont be
    set up for it), although you may be able to RTP compression.

    if you are using an IPsec internet VPN as well, then that will add a fair
    bit of padding.
    >
    > Here's the main questions:
    >
    > How's the system likely to cope with all 30 sites in a conference over
    > the current/upgraded ADSL line?


    ignoring QoS and background traffic - 10 calls on the current link.

    In reality, if there is any other traffic you should derate this - the"rule
    of thumb" i use on Cisco call manager setup is based on cisco
    recommendations.
    it is 1/3 of the bandwidth for all the real time (in a corporate general
    purpose net) - this assumes you are using QoS and you have other traffic on
    the same links.
    >
    > I understand that the choice/mix of codecs used when conferencing can
    > have a big impact on the results - our current phone of choice supports
    > the usual a-law and u-law stuff and others will also do ilbc and gsm -
    > obviously, the latter two use less bandwidth but what is likely to
    > happen if we mix them with phones on a-law/u-law or should we just stick
    > to one set of common codecs - which would be the bigger bandwidth
    > a/u-law ones?


    a lot depends on what you want - but i much prefer G.711 conferencing.
    >
    > I have seen some dev threads + others on silence suppression and keeping
    > it OFF in order to provide the proper timing for RTP streams - does this
    > still apply with A@H 2.8?


    Not sure - but does it help?

    silence suppression is based on the assumption that each line is "off" 50%
    of the time in each direction for typical conversations.
    But with a peer to peer conference the outbound link carries a "mix" of all
    input - so it is going to be on most of the time. And outbound bandwidth is
    where your bottleneck is.
    >
    > How does the ADSL bandwidth relate to max number of simultaneous
    > connections? I can appreciate a simple calculation of (available
    > bandwidth / codec bandwidth requirement), but guess in the real world
    > it's not as simple as that when considering things such as simultaneous
    > use of channels not actually transmitting/receiving 100% of the time,
    > but how aboout the streams associated with conferences and also the
    > 'overhead' of transmitting all the silence?
    >
    > I have had a good look around the main Asterisk and Asterisk@Home sites
    > and have found some generalised info on all of this, but not a lot of
    > real world input.
    >
    > Any insight appreciated.


    a lot depends on how your network is glued together and the implications
    that has for actual useful VoIP bandwidth (and all quality issues such as
    latency and loss rates)

    if you expect to run 30 user conferences, then more bandwidth outbound. You
    either need a better link or something symmetric such as SDSL.

    Maybe 1 way to cut down the cost of bandwidth is to have the server hosted
    somewhere? you usually get 10M or more then.
    >
    > Thanks

    --
    Regards

    - replace xyz with ntl
    stephen, May 21, 2006
    #2
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  3. linker3000

    linker3000 Guest

    stephen wrote:
    > "linker3000" <> wrote in message
    > news:44708a9c$0$8335$...
    >> Hi guys,
    >>
    >> I posted this in the A@H forums but hope I might catch a few more
    >> helpful people here too...
    >>
    >> I'd appreciate some real-world experience if possible:
    >>
    >> If we roll out our * trial, we will initially have about 30 IP phones
    >> (no analogues), one per branch office, and I was wondering about
    >> conference/bandwidth issues...
    >>
    >> The server is a dual core P4-2.6GHz with 1GiB RAM so I guess we have
    >> enough steam to start with?! At the moment the server is conncted to a
    >> 2Mbit ADSL line with an upstream speed of 288Kb. I will probably upgrade
    >> this to 8Mbit/448Kb service.

    >
    > i dont know the asterisk specifics - but your link is going to limit the no.
    > of simultaneous calls.
    >
    > depending on the codec you use, you are going to need 80 Kbps or more for
    > G.711, or 24 Kbps for G.729.
    >
    > you cant have header compression on the link (since the ISP routers wont be
    > set up for it), although you may be able to RTP compression.
    >
    > if you are using an IPsec internet VPN as well, then that will add a fair
    > bit of padding.
    >> Here's the main questions:
    >>
    >> How's the system likely to cope with all 30 sites in a conference over
    >> the current/upgraded ADSL line?

    >
    > ignoring QoS and background traffic - 10 calls on the current link.
    >
    > In reality, if there is any other traffic you should derate this - the"rule
    > of thumb" i use on Cisco call manager setup is based on cisco
    > recommendations.
    > it is 1/3 of the bandwidth for all the real time (in a corporate general
    > purpose net) - this assumes you are using QoS and you have other traffic on
    > the same links.
    >> I understand that the choice/mix of codecs used when conferencing can
    >> have a big impact on the results - our current phone of choice supports
    >> the usual a-law and u-law stuff and others will also do ilbc and gsm -
    >> obviously, the latter two use less bandwidth but what is likely to
    >> happen if we mix them with phones on a-law/u-law or should we just stick
    >> to one set of common codecs - which would be the bigger bandwidth
    >> a/u-law ones?

    >
    > a lot depends on what you want - but i much prefer G.711 conferencing.
    >> I have seen some dev threads + others on silence suppression and keeping
    >> it OFF in order to provide the proper timing for RTP streams - does this
    >> still apply with A@H 2.8?

    >
    > Not sure - but does it help?
    >
    > silence suppression is based on the assumption that each line is "off" 50%
    > of the time in each direction for typical conversations.
    > But with a peer to peer conference the outbound link carries a "mix" of all
    > input - so it is going to be on most of the time. And outbound bandwidth is
    > where your bottleneck is.
    >> How does the ADSL bandwidth relate to max number of simultaneous
    >> connections? I can appreciate a simple calculation of (available
    >> bandwidth / codec bandwidth requirement), but guess in the real world
    >> it's not as simple as that when considering things such as simultaneous
    >> use of channels not actually transmitting/receiving 100% of the time,
    >> but how aboout the streams associated with conferences and also the
    >> 'overhead' of transmitting all the silence?
    >>
    >> I have had a good look around the main Asterisk and Asterisk@Home sites
    >> and have found some generalised info on all of this, but not a lot of
    >> real world input.
    >>
    >> Any insight appreciated.

    >
    > a lot depends on how your network is glued together and the implications
    > that has for actual useful VoIP bandwidth (and all quality issues such as
    > latency and loss rates)
    >
    > if you expect to run 30 user conferences, then more bandwidth outbound. You
    > either need a better link or something symmetric such as SDSL.
    >
    > Maybe 1 way to cut down the cost of bandwidth is to have the server hosted
    > somewhere? you usually get 10M or more then.
    >> Thanks


    Stephen,

    Thanks - that kinda backs up my thoughts and I was actually scoping out
    SDSL and hosted services this afternoon.

    L3K
    linker3000, May 21, 2006
    #3
  4. linker3000

    MartinC Guest

    The half way house might be the Zen Office 8000 Max service for £79 ex
    VAT.
    With downstream speeds of up to 8Mbps and upstream speeds of up to
    832Kbps.

    Martin
    MartinC, May 24, 2006
    #4
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