Advice on net/asterisk

[Adam Vande More]

On Tue, Mar 3, 2015 at 12:55 PM, Janos Dohanics <web at 3dresearch.com> wrote:

> On Tue, 3 Mar 2015 11:19:30 -0600
> Adam Vande More <amvandemore at gmail.com> wrote:
>
> > On Tue, Mar 3, 2015 at 9:07 AM, Janos Dohanics <web at 3dresearch.com>
> > wrote:
> >
> > > Hello List,
> > >
> > > I'm considering to use net/asterisk for an office with 10-12 phones.
> > > I'd welcome your suggestions, advice, tips, comments...
> > >
> >
> > Are you planning on PRI or SIP?
>
> I'm not sure how to make an educated decision on this, that is, if we
> could get by with SIP. I tried http://www.voipqualitytest.com/ and this
> is what it shows:
>
> 1. Random time during work day:
>
> Speed test statistics
> ---------------------
> Download speed: 10537 kbps
> Upload speed: 1115 kbps
> Download consistency of service: 64 %
> Upload consistency of service: 67 %
> Download test type: socket
> Upload test type: socket
> Maximum TCP delay: 118 ms
> Average download pause: 1 ms
> Minimum round trip time to server: 54 ms
> Average round trip time to server: 57 ms
> Estimated download bandwidth: 13651 kbps
> Route concurrency: 1.2954974
> Download TCP forced idle: 33 %
> Maximum route speed: --
>
> VoIP test statistics
> --------------------
> Jitter: you --> server: 1.1 ms
> Jitter: server --> you: 24.5 ms
> Packet loss: you --> server: 0.0 %
> Packet loss: server --> you: 0.0 %
> Packet discards: 0.0 %
> Packets out of order: 0.0 %
> Estimated MOS score: 3.6
>
> 2. Same, after introducing additional outgoing traffic:
>
> Speed test statistics
> ---------------------
> Download speed: 10920 kbps
> Upload speed: 705 kbps
> Download consistency of service: 85 %
> Upload consistency of service: 51 %
> Download test type: socket
> Upload test type: socket
> Maximum TCP delay: 71 ms
> Average download pause: 1 ms
> Minimum round trip time to server: 54 ms
> Average round trip time to server: 87 ms
> Estimated download bandwidth: 12194 kbps
> Route concurrency: 1.116652
> Download TCP forced idle: 51 %
> Maximum route speed: --
>
> VoIP test statistics
> --------------------
> Jitter: you --> server: 8.7 ms
> Jitter: server --> you: 3.6 ms
> Packet loss: you --> server: 2.0 %
> Packet loss: server --> you: 0.0 %
> Packet discards: 0.0 %
> Packets out of order: 0.0 %
> Estimated MOS score: 2.5
>
> 3. And with incoming traffic:
>
> Speed test statistics
> ---------------------
> Download speed: 9693 kbps
> Upload speed: 1465 kbps
> Download consistency of service: 48 %
> Upload consistency of service: 66 %
> Download test type: socket
> Upload test type: socket
> Maximum TCP delay: 122 ms
> Average download pause: 1 ms
> Minimum round trip time to server: 59 ms
> Average round trip time to server: 87 ms
> Estimated download bandwidth: 12446 kbps
> Route concurrency: 1.2840027
> Download TCP forced idle: 51 %
> Maximum route speed: --
>
> VoIP test statistics
> --------------------
> Jitter: you --> server: 0.5 ms
> Jitter: server --> you: 13.6 ms
> Packet loss: you --> server: 0.0 %
> Packet loss: server --> you: 0.0 %
> Packet discards: 0.0 %
> Packets out of order: 0.0 %
> Estimated MOS score: 3.9
> I guess a MOS score of 2.5 is below what's acceptable, but please
> comment.
>
> --
> Janos Dohanics
>

Depends on your use case.  If phone is mission critical to the business and
required to be highest quality all the time, then PRI is pretty much going
to be required.

With SIP, it's generally much cheaper trunking, scaling, and provider
switching, but you're much more susceptible to network issues.  So if you
don't have a good full QoS/CoS setup and aren't going to implement one,
along with have rock solid and maybe dual homed pipes then SIP might not be
a good choice.

-- 
Adam