[Ninux-Calabria] I: [Ninux-Wireless] Asterisk Distribuito, no solo dentro Ninux ma globalmente

[Giuseppe De Marco]

rigiro la mail discussa ieri in riunione

----- Messaggio inoltrato -----
Da: Giuseppe De Marco <peppelinux a yahoo.it>
A: "wireless a ml.ninux.org" <wireless a ml.ninux.org> 
Inviato: Martedì 9 Luglio 2013 12:45
Oggetto: Re: [Ninux-Wireless] Asterisk Distribuito, no solo dentro Ninux ma globalmente
 


> scarica questo libro:

> http://it-ebooks.info/book/2332/


è una bibbia. Su asterisk avevo due libri ma nessuno supera questo.

Molto interessante dundi, sebbene vada studiato più approfonditamente dal sottoscritto (se non lo vedo funzionare...).

Se il protocollo da adottare in Ninux è SIP, chiedo in lista se qualcuno ha avuto esperienze su SipX (sipXecs). Questo supporta esclusivamente il protocollo SIP e nasce per essere facilmente scalato per mezzo di proxy.

"""
a) sipXecs IP PBX uses external gateways. It supports as many external gateways you need without limit and offers automatic failover in case a gateway is unavailable or busy. It also offers least cost routing where gateways can be deployed anywhere you need them.

b) sipXecs IP PBX does not route calls (media) through the server because it separates signaling from media. Therefore, sipXecs can support as many simultaneous calls as your LAN / WAN bandwidth permits. Asterisk has a hard limit because calls go through the Asterisk server. For a dual-core XEON with 2 GB RAM that limit is 60 simultaneous calls. 

c) Routes calls direct peer-to-peer and not through the call control server. The sipXecs IP PBX also supports HD voice for both end points as well as PBX services such as conferencing, voicemail and auto-attendant. Video support poses no additional load on the sipXecs IP PBX as the media streams do not go through the PBX server.

Asterisk is modeled like a more traditional TDM PBX where lines ("channels") come into the PBX that carry voice and signaling. The IAX protocol even explicitly bundles voice and signaling along the same route. This not only uses more bandwidth than necessary, imposes additional delay, injects additional jitter, and represents a single point of failure, this also severely limits the total number of calls that can go on at any given time for the system. 

"""


Idee, intuizioni, opinioni: sono necessarie.

[1] http://www.voiptoday.org/index.php?option=com_content&view=article&id=63:sip
[2] http://wiki.sipfoundry.org/pages/viewpage.action?pageId=9929235
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