Hope things are going well for you. My comments are below.
Cuneyt M <[email protected]> 03/31/09 5:35 PM >>>
Hi Tony,
Good to hear from you.
I didnt know only 3.11.x and 4 will have native SIP trunking, Do you
know when is the realistic release date for 4?
**No. I am checking the tracker to see what kind of issues are left, but there
is also a lot of activity. I would hope sometime in April, but I reserve the
right to re-guess that too.
So for that kind of volume you need a minimum of 15 trunks and a 20-pack of DID numbers.
20-pack DID numbers because they come in 20? in your example we'll be
using 15 trunks=lines where each will have a DID. What would be the use
case for the additional 5 DIDs?
**20 pack of DID's because it's cheaper than 15 when you buy in bulk. If all of
your agents are on the phone at once then you will need at least 15 trunks. The
question comes when you get a busy signal for incoming calls because you don;t
have enough trunks. Bandwidth.com gives you the ability to add trunks later,
but their paperwork process takes about a week to go through provisioning. You
can always add lines, but you'll want to try to be accurate to start with.
Remember to spare some bandwidth for this, *on tten simultaneous voice
conversations 9where audio is playing of any sort, not ringing, like leaving or
listening to voicemail, AA or speaking interactively with another person, then
conservatively you need to budget roughly 800k for bandwidth in
upload/download, *
**We tracked SIP calls here from a Polycom phone on our network through a DPI
system. It showed us marvelous things, and we watched in amazement as some
calls tipped 80k of bandwidth as I recall. When we ran 5 calls simulaneously
through bandwidth.com, we saw somewhere around 400+k of bandwidth being used.
So in real case 10 calls needs 800k, 15 needs 1200k in both directions.
I didnt quite get your point here. Can you please re-phrase it within
actual use-case scenario. You clearly have experience on getting this up
and running and dont wanna miss what you are saying in the
red-highlighted sentence.
Regariding blue-highlited sentence, budgeting 800k bandwidth for both
upload and download is calculated on 15 concurrent user basis? i.e. How
do you calculate the bandwidth requirement as 800k? Is this an
estimation based on G711u/a codec (if i do recall correctly g711
required 64Kb per line but i could be mistaken) ?
If the calling end is Philippines and receiving end is
US(Bandwidth.com), taking the distance and possible-hops between two
locations which codec usage will be logical?
And how should I calculate the bandwidth requirement accordingly?
Bandwidth.com offered a 30 day trial (not sure yet if its free or with a
fee) and i was thinking i can play with it in SipX 3.10.3 by adding
under Gateways.
**It wont work with a sip aware firewall or trunking appliance, free or not.
I don't have any test system as all live systems are running 3.10.3 at
the moment and i saw quite a number of issues reported for the 3.11.x
which scared me to update any of the live systems. As we are on tight
budget, is there any other way to run Bandwidth.com SIP Trunking without
Ingate - like an opensource software that runs on sipx or other system?
I heard about OpenSBC, SIPproxyD but didnt have the chance to look at
them and what problems they solve.
Does any of these or other open source software are able to make SipX
3.10.3 SIP Trunking work, without Ingate sort of appliance?
**I don't think siproxd is worth any effort. OpenSBC shows some promise but it
seems a lot to do and is somewhat finicky. I tried twice and gave up citing not
documented enough yet and the forum was not very helpful unless I wanted to buy
a support agreement, at which point I went and got an Ingate, learned that, and
then cycled through a couple of problematic ITSP's before coming to
Bandwidth.com.
If it were me, and it's not, Ingate would be a sure bet and would not require
any changes inside your network and would seamlessly get 3.10.3 working with
bandwidth.com. If you need to wait for 4.0, I suggest a test system gets setup
so you can work your way into it, as sipXconfig is much changed and there seems
to be a bit of a learning curve for me while I understand the nuances of
sipXbridge and sipxrelay.
Thank you for your prompt reply and very informative e-mail.
all the best!
Tony Graziano wrote:
On 3/31/2009 at 4:38 PM, in message <[email protected]>, Cuneyt M
<[email protected]> wrote:
Hello everyone,
We have existing setup of SipX 3.10.3 with AudioCodes gateways.
However, we will be testing/integrating Bandwidth.com US Sip-Trunk (with
DID) with SipX. I understand that we can add a Sip Trunk just like
another gateway in Sipx(?)
In 4.0 the sipx system is able to communicate with ITSP's, there is already a
preconfigured template for Bandwidth.com in the current 3.11x development
version. In 3.10.x you will still need a sip trunking device, like an Ingate.
The main use-case would be Philippines satellite-office with local SipX
server routing +1 numbers via the Bandwidth.com's SIP Trunk.
I assume we'll need one Sip Trunk per extension (if we are to assume
each extension will be sales guys calling their leads in US) - correct
me if im wrong.
Think of "trunks" as "lines". These are call paths. If you have a two-way trunk (inbound
and outbound calling), then you can either receive or place a call on the trunk (line). If you have 20
people, and 5 are making calls and 5 are on calls that came in and 5 more are in the process of ringing,
that's "15" simultaneous call paths. So for that kind of volume you need a minimum of 15 trunks and
a 20-pack of DID numbers. Remember to spare some bandwidth for this, on tten simultaneous voice conversations
9where audio is playing of any sort, not ringing, like leaving or listening to voicemail, AA or speaking
interactively with another person, then conservatively you need to budget roughly 800k for bandwidth in
upload/download, and if you are sending it through a firewall you might look at prioritizing your traffic
based on destination and port.
We have no prior dealing with ITSP or SIP trunking as we used AudioCodes
GWs for local or close-proximity termination.
Can you please share your thoughts and experience in setting up a Dial
Plan in SipX 3.10.3 where it'll route the call to the registered
ITSP(under Gateways) for numbers that start with +1 (US numbers).
I would think that the number would choose the gateway so to speak, seems
straightforward.
Also if you can share your experience on US-based Sip Trunk
providers(Bandwidth.com and others) for service reliability, features
and integration with SipX 3.10.3 i would very much appreciate.
We use them now with an Ingate, it's been very stable and pleasant, and the config for
the Ingate is one of the least complicated I've seen. I do think you need a static IP
address for bandwidth.com, and it makes registration issues a "non-issue".
Thanks in advance.
all the best!
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