[asterisk-ss7] Where's Matt been? Well, here's the explanation
Matthew Fredrickson
creslin at digium.com
Thu Mar 25 23:40:29 CDT 2010
Hey all,
It's been a long time. I apologize for my quietness for the last while
here, it has been a very busy year this last year with some of the other
projects I've been working on.
I just wanted to let everyone know that I'm still alive, and haven't
given up or forgotten about libss7, and Asterisk with SS7.
I appreciate very much Attila's great work on getting libss7 polished
up. He's done a very good job taking it to the next level, and has done
a great job helping everyone out. Thanks very much Attila.
As far as getting his code merged back in, I was in the midst of this a
while ago, but had to stop due to personal time issues as well as some
other potentially architectural issues that subsequently came up.
Hopefully we can get that moving again in the next little bit though.
I actually have been quite busy, and hopefully have some things to show
for it, two things actually.
I'll not go into too much detail tonight (as it's getting quite late)
but I'm looking for some hungry testers, that wouldn't mind beating up
on some probably alpha level code.
They are:
1.) libss7 point code clustering support.
Basically, you can have Asterisk boxes share signalling links now using
this code. Although the signalling links are physically terminated on
other machines, you can plug bearer T1s/E1s into other Asterisk boxes
and virtually utilize the signalling links of the other boxes.
2.) A new channel driver, called chan_ccs, that allows, among other
things, you to control MGCP media gateways for bearer trunks, instead of
having to locally terminate them on the asterisk box that's controlling
the signalling links. There is also code in the same branch that has
chan_ccs that modified chan_mgcp so that Asterisk can act as a media
gateway (since I don't have any good real media gateways to test on).
This basically means you can have Asterisk TDM channel scalability up to
(in the ideal state) the same level as you can do with SIP with no
media, per box.
In essence, this is turning Asterisk into a "true" softswitch, allowing
native bridging between media gateways and any other RTP endpoint
(including other media gateways). This also means that you don't have
to terminate bearer T1s/E1s on the main signalling box.
-- So, what does this mean for you, you may be asking?
These are really the next steps in making big SS7 work with Asterisk.
They both allow for scaling a point code across multiple asterisk
machines, and distribution of bearers on different boxes than the ones
that contain signalling links.
Like I said though, most of the work is in a functional but early phase,
and so I need some people that are interested enough in the added
functionality that they're willing to work with potential hickups along
the way.
Some of the changes I've had to make to libss7 have made it further more
difficult to merge Attila's changes back in, which is the other reason
why it has been so long and it still has not been merged.
If you're interested, either reply to me or this thread and let me know.
Thanks again,
Matthew Fredrickson
Digium, Inc.
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