Hi Skyler,
I guess the first big decision is about the architecture:
1) fully centralized - phone from all location do register and send
calls to a single SIP server (in a master location) (as Mark suggested)
Advantages : simple arch, easy to put in place, few components, easy to
provision and collect CDRs (a single point)
Disadvantages : if one site loses the IP connection, it will be
completely down (not even in-site calls will work)
2) distributed arch - your idea, of placing a small IP-PBX in each
location do handle in-site calls and a master sip server to connect all
sites and to do PSTN termination.
Advantages : a site may survive (as internal calls) even if disconnected,
Disadvantages : complex arch, multiple components (PBXs, servers),
complex provisioning.
So, depending on + or - for each approach, you have to choose the best
way for you.
Regards,
Bogdan
Skyler wrote:
Mark -
Thanks for sharing your thoughts, they are definitely helping to put
the pieces of this puzzle together. Today I spent most of the day
mapping out each office via the net and found the common backbone
interconnects. At these x-connects I found 2 data centers. All offices
are 30-40ms from one or the other and both DC's are 15-20ms from each
other. I couldn't figure out what the distance would be from the DC to
the provider, though I know the provider is in a major DC and one
Province over so it can't be more than 15-20ms across the backbone.
Both DC's offer dedicated servers, so we are going to look into
putting one server at each DC and ditch the original regional/national
plan for a more conservative and easy to manage plan. I'm confident
now that there will be better overall quality going this way.
Now its time to unscramble the mess that is my install notes and
document a clean OpenSIPS+Asterisk install before moving further.
After that I'm a bit lost though as I know that we need NAT but not
sure which solution is best / easiest to work with (RTPproxy,
NAThelper, MediaProxy). From what I've read up on each, Nathelper
seems to be built into Osips whereas RTPproxy and MediaProxy require a
possibly troublesome install vs loading module/adding code. Searching
the mailing archives hasn't been enough for me to decide on a winner.
From what it sounds like, you have a lot of experience in the setup
that I'm working on building. Out of curiosity, which method do you
prefer for resolving far-end NAT issues?
Skyler
On Mon, Jan 17, 2011 at 1:26 AM, Mark Sayer <[email protected]
<mailto:[email protected]>> wrote:
Skyler -
We are a South Pacific regional provider of hosted PBX services so I
may be prejudiced toward a like infrastructure. Some of our customers
are 3000kms from our servers but the ping times are still less than
50ms so I'm curious why yours are so long. That said, 200ms is sort of
the magic number you don't want to exceed. (Having said that, we do
get some pretty decent call quality connecting to some terminators who
are over 250ms away. 50+250 and its still OK.) Call quality is 99%
Internet connection. OpenSIPS + Asterisk works perfectly with every
call but if the Internet (which you can't control) plays up you get
flack for providing a bad service.
I'd recommend spending some time looking at your Internet connections.
Can you get them all from the same provider? (I don't even know what
sort of connections you are talking about. We actually get business
grade voice quality from ADSL over copper.) Can you locate your server
in a data center that has good connections to both your ISP and your
terminator? My dream has always been to have a large rack of equipment
in the back office but to make our service work I've had to locate in
a major data centre hundreds of kms away. Our office isn't nearly as
impressive as our service is but that's what the customers pay for.
I'd only put servers in the offices if there was some reason that
functionality was needed there. Even if you need a receptionist at
each office that can all be handled from a single Asterisk box.
Just more thoughts.
Mark
On Mon, Jan 17, 2011 at 6:11 PM, Skyler <[email protected]
<mailto:[email protected]>> wrote:
> Hi Mark,
> Thanks for the reply. So if I understand correctly, I am
thinking too big.
> K.I.S.S as some say.
> The existing PBX's are extremely old, so breakdowns & phones are
a problem
> and we don't want to repair anymore. In the suggested scenario
would you
> recommend replacing the existing hardware (as they breakdown)
with IP phones
> and Asterisk at each office then or just ditch the Asterisk and
have all the
> phones register to OpenSIPS directly at HQ? My concern is call
quality with
> 110ms to HQ then 75ms to provider = 185ms from furthest office,
is this
> still not an issue?
> Thanks,
> Skyler
>
> On Sat, Jan 15, 2011 at 4:55 PM, Mark Sayer
<[email protected] <mailto:[email protected]>> wrote:
>>
>> Here is one suggestion:
>> - single OpenSIPS & Asterisk at central office
>> - use Asterisk as gateway to PSTN (for all offices)
>> - connect remote office PBXs to central office using using
multi-port
>> FXS gateways
>> - 110ms is no problem
>> - single system admin point, single cpu, 200 or more concurrent
calls
>> - no admin, low cost at remote offices
>>
>> Mark
>>
>
> _______________________________________________
> Users mailing list
> [email protected] <mailto:[email protected]>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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