The Asterisk Developer Team is proud to announce the Asterisk SPE
v1.0 Beta Release
for immediate download on tftp.digium.com.
The SPE has been developed as a joint project between Digium, the
Asterisk Company,
Voop, the European Asterisk Dialtone provider and the Asterisk
community.
The Asterisk Service Provider Edition is focused on the needs for the
new breed
of Telecom companies - the Voice over IP Service Providers. It will
be available
both as a free download in Open Source and as a commercial product
called Asterisk Commercial Service Provider Edition, ACSPE.
- We felt the need to focus on being an enabler for this new kind of
telco,
making sure that Asterisk fits into their network as well as
business models
in a professional way says Matt Penser, Asterisk innovator. The
previous versions
was more targeted to the needs of the business user, a market where
Asterisk already is stronger than any other offering on the market.
The Asterisk SPE has a number of new features, that makes it the most
powerful
platform for these companies. No other Open Source package can deliver a
matching feature set:
- All the features from Asterisk 1.4 and the business edition
- Asterisk VoipRoute(R) technology for SmartRTP(R) bridging
- Asterisk RateRoute(TM) technology for route selection
- Asterisk SpitWall(R) core for SPIT filtering
These new solutions will enhance Asterisk and will help the VSP's to
leap lightyears ahead of their competion.
* Asterisk VoipRoute(R) SmartRTP(R) Bridging
The VoipRoute SmartRTP bridging technology enhances the Asterisk RTP
bridge with a new scheme. In addition to the current RTP bridges -
the native bridge,
the remote bridge and the hybrid RTP-direct bridge, SmartRTP uses a
combination
of the BGP IP routing protocols and the TRIP VoIP routing system to
find the
best and fastest way to route calls between IP nodes on the Internet
or local network.
- The SmartRTP bridge system, based on our patented VoipRoute core,
makes sure that call latency is minimal. We also enhanced it with a
MediaRescue solution that will capture lost media frames and re-
insert
them in the audio or video stream before it reaches the
destination. says
Josua Polk, the Asterisk RTP developer.
This system implements an Asterisk VoipRoute layer on top of the
Internet
and uses Dundi(TM) to automatically discover new SmartRTP relays
and their
properties. It practically erases packet loss, jitter and latency
from the list of
issues for the provider's support department. We call it SPEake-
friendly calls!
* Asterisk RateRoute(TM) Least Cost Routing
-
The RateRoute(TM) solution is only available in the ACSPE due to
licenses from
other vendors, soon to be disclosed. The RateRoute system analyze the
call
from fifteen distinct properties and use an external hardware
accelerator to
find the best route to forward the call, be it PSTN or VoIP channels.
By using
the hardware accelerator RR520P PCI express card, LCR decisions is now
down to microseconds without accessing external databases.
- We've implemented this in our commercial VoIP network during
development,
and cut our costs by at least 75% and enhanced call quality.
Billing and CDR
mediation is much easier, since the RateRoute system always
picked one
outbound service provider that always matched the fifteen
criteria for
carrier selection says Anders Runnstam at PulseVoip in Bergen,
Norway.
* Asterisk SPITwall(R) - filtering away tomorrows VoIP spam today!
--
The SPITwall(R) technology is developed by Olle E. Johansson, a
member of the Asterisk developer team and Senior Technical Advisor
for Voop in Bergen, Norway - the Asterisk Dialtone provider.
- I got more and more annoying calls during development, which
disturbed
me a lot and caused me to loose concentration. On the other hand,
it inspired me to develop SPITwall to be able to filter them out.
I have measured up to 95% success rate on call filtering,
which is far beyond any similar products on the market. By not
bothering
with answering the final 5%, I could concentrate on development again
and succesfully finish my development projects. says Olle.
The SPITwall is built on a shared database and use bayesian techniques
to analyze the content of the call. It requires Asterisk ChanSpy to
be able
to listen in and warn the callee about ongoing unsolicited calls. The
callee
can also press certain DTMF sequences during the call to mark the call
as SPIT. The voice pattern, SPITwall checksums and call properties will
then immediately be stored in the Digium SPITcore repository to be
available for all other users.
- Using the community to build a SPIT-fighting
