[asterisk-users] Viable using purchasing sip lines

Stephen Bosch posting at vodacomm.ca
Mon Apr 30 09:18:30 MST 2007


I can try and answer some of your questions.

kenny.kant at running-config.com wrote:
> Hello All,
> We have been doing Asterisk and CME implementations recently but we
> almost always exlusively bring in analog lines and or PRI for PSTN
> access to our systems.  I have known about providers providing SIP based
> lines and SIP trunks to end users for PSTN access.  I am curious about
> the following:
> - How practical is this?  The idea of terminating pstn calls to across
> the Internet which is an unguarenteed medium concerns me.  Even if our
> access to it is quazi stable T1 data type of access.  Do any of you do
> systems where this is soley the method used for incoming calls from the
> pstn?  If this is done are there things to look for in a SIP provider,
> as in their presence on the Internet latency ..etc?

You're absolutely right to be concerned. If you need critical service,
leave the VoIP terminations for the time being.

We *do* use it as an adjunct to primary PSTN lines; so, for example, we
might dump a bunch of extra PSTN lines and replace them with SIP or IAX
connections; in one case, we just "forward on busy" from the main PSTN
number to the SIP numbers, and that works quite well. When it's working
users even like the sound quality better. When it's working ;)

It does work most of the time, but when it comes to phone service, user
expectations are way different (hell, my expectations are way different,
so I can understand).

Here are the things I would recommend you pay close attention to when
choosing a SIP provider:

1. There are numerous fly-by-night operations. Providing a stable, high
availability service is not trivial and costs some money; you want a
provider that offers as close to round-the-clock support as you can get.
One of the best tests is just to call the provider directly and see what
kind of experience you have. When calling some SIP carriers, I have had
dropouts, chirping, clicking, calls terminating nowhere, etc. Those are
the ones you want to avoid :) Also, if you are spending a long time in
queue... be suspicious. There are outfits run by "two guys in a
basement" -- and those two guys are often away skiing or windsurfing,
depending on the season. Just be vigilant.

2. Ask for the IP of the PSTN POP for the provider and check the
latency. I wouldn't tolerate anything higher than 75 ms, and shoot for
something under 50 ms if you can manage it. Sometimes it's not possible
and depends where you're located.

> - What are the major advantages?  I know some places provide all you can
> eat plans which could be seen as a plus and some others provide really
> low rates. Are there others?

The major advantage is that, in general, it is easier to support more
channels (when you need them) and the cost per channel is, on average,
lower than for a PSTN connection. For example, with certain providers,
it's possible to get two channels for slightly more than the cost of a
single business line with the PSTN provider.

Another advantage: if you only need a few channels, and your PSTN lines
are analog, then VoIP connections offer you some call progress
detection, which can be useful when you're trying to do "follow me"
ringing off the premises.


More information about the asterisk-users mailing list