VoipExperts

O time de desenvolvimento do Asterisk está orgulhoso em anunciar o Asterisk SPE (Service Provider Edition) v1.0 Beta disponível para download em tftp.digium.com. O SPE foi desenvolvido como um projeto conjunto entre a Digium, a Voop, um provedor europeu e a comunidade do Asterisk.

O Asterisk Service Provider Edition está focado nas necessidades para o novo ramo das companhias de Telecom - os provedores de Voz sobre IP. Estará disponível tanto para download free como um produto comercial chamado Asterisk Commercial Service Provider Edition, ACSPE.

"Nós sentimos a necessidade de focar em ser um provedor para esse novo tipo de telco, tendo certeza de que o Asterisk se encaixa nas suas redes bem como modelos de negócios de uma forma profissional. A versão anterior estava mais direcionada para as necessidades do usuário em negócios, um mercado onde o Asterisk já é mais forte do que qualquer outra oferta do mercado.", disse Mark Spencer, entusiasta do Asterisk.

O Asterisk SPE tem um bom número de novas funcionalidades, que faz dele a plataforma mais poderosa para essas companhias. Nenhum outro pacote Open Source disponibiliza as seguintes opções:

- Todas as funcionalidades do Asterisk 1.4 e business edition
- A tecnologia Asterisk VoipRoute(R) para bridging SmartRTP(R)
- A tecnologia Asterisk RateRoute(TM) para seleção de rota
- Núcleo Asterisk SpitWall(R) para filtro de SPIT

Essas novas soluções irão aprimorar o Asterisk e irão ajudar as VSPs a pular anos luz à frente dos concorrentes.

* Asterisk VoipRoute(R) SmartRTP(R) Bridging
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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
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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!
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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 lose 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 database is natural for an Open Source project. The community is the power of Asterisk and by sharing a resource like this, we can make sure that everyone contributes. The SPITshare(r) analyzer makes sure that companies that does not contribute will get older data and more SPIT calls" says Jill Timmer, VP or marketing.

SPITwall 1.0 is available with English, Norwegian and Swedish language support. Some support for Canadian and southern US dialects is implemented, and will be finished by release time.

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In addition to these revolutionary features, Asterisk Service Provider Edition will contain full T.39 faxing (leapfrogging systems that only support up to version 38), the Codename Pomengranade SIP Stack, the Project Okapi IAX3 trunking technology featuring IAX3 over SS7 transport, the VoipVote digital phone voting system with app_preselect cheating technology and a fully working version of the VirtuAST virtual PBX hosting Asterisk virtualisation core.

Asterisk SPE v1.0 beta is available for immediate download. At this time we're looking for feedback from service providers.

Release date for the 1.0 version is to be released, pending beta tests. ACSP Edition will be available with Telco level 24/8 support May 15th, 2007. The RR520 RateRoute hardware accelerator is in distribution through authorized resellers starting April 10th.

Asterisk, Digium, SPITwall, SpitShare, VoipRoute, SmartRTP, RateRoute and Dundi are trademarks that may be registered by Digium, Inc.

For immediate release, April 1st 2007 On behalf of the Asterisk development team and project 0401

Fonte: http://www.sineapps.com/news.php?rssid=1717