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the end more on University dumps Cisco VoIP for open-source Asterisk
From: David Farber <dave () farber net>
Date: Wed, 3 Jan 2007 11:57:45 -0500
Begin forwarded message: From: Michael Slavitch <michael.slavitch () e2exchange com> Date: January 3, 2007 11:51:56 AM EST To: dave () farber net Cc: ip () v2 listbox com, ferber-ip () glomph com, hsinnrei () adobe comSubject: Re: [IP] more on University dumps Cisco VoIP for open-source Asterisk
As opposed to barefaced promotion of Asterisk from Tim O'Reilly? Is Asterisk above scrutiny? Really? Why?
Asterisk isn't even an open SIP switch, it is a VOIP PBX that must translate SIP into its own media format, and is highly biased towards the use of proprietary hardware from Digium.
On the other hand SIP is an IETF standard signalling protocol that is independent of media, this allows heterogeneous distributed systems to be built using components that follow open standards, whether commercial or open source. Open source is useless if it demands another silo, and a religious devotion to it does just that. Commercial software that correctly implements SIP offers more flexibility than closed architectures, be they free software or commercial hardware, especially if the open source system adopts religion and cannot accept criticism.
The point my message is that the advantage does not lie with free software but with standard protocols. Once open standards start operating in scale proprietary systems whither and are replaced with the best system for the job, whether open source or commercial.
Regards Michael PS: Here is some detail about Asterisk. From Wikipedia:"Unlike most next generation telephony applications and architectures, Asterisk is built using a monolithic model and does not break out individual parts of the application. An alternative application architecture would be disaggregated, where each architectural component could exist independently and therefore be scalable or easily accelerated with best of breed components. As an example, the media server portion of the Asterisk application is not implemented as a Voice Over IP media server and cannot be easily replaced with a hardware-based alternative. The Asterisk architecture therefore has inherent scalability problems in terms of raw performance and functional completeness. Due to the dual licensing nature of the Asterisk code and the feeling among some developers that changes required to the software were not incorporated in a timely manner, the Asterisk source base was forked in 2005.
Asterisk's timing mechanism (both timestamps and wake-ups) is strongly biased toward systems which contain at least one of Digium's PCI boards. Users with other needs, including pure VoIP setups, are likely to experience timing problems. The problem can be somewhat reduced by loading a kernel module which emulates the timing functionality normally provided by Digium's PCI boards. As with the other Digium drivers, this driver has not been submitted for inclusion in the standard Linux kernel and thus presents the usual awkwardness associated with non-standard drivers."
While Digium's IAX and IAX2 are free open source protocols they are not open standard protocols independently maintained by a standards body, they are indeed proprietary.
On 1/3/07, David Farber <dave () farber net> wrote: Begin forwarded message: From: Jerry Glomph Black <ferber-ip () glomph com> Date: January 3, 2007 11:23:38 AM EST To: David Farber < dave () farber net> Subject: Re: [IP] more on University dumps Cisco VoIP for open-source Asterisk Dave, this long posting from Mr Slavitch is a barefaced commercial promotion, and like most commercial competitors to Asterisk, must resort to distortion (or worse) about Asterisk. Not quite as bad as what I had to suffer in a meeting with Cisco reps last week, but not that different in its tone. Where does he get off on saying 'Asterisk is not based on open standards'? Asterisk supports more VoIP protocols than any other such product. You can run it purely as a SIP creature should one choose to do so, or any combination of other protocols simultaneously. _________________________________________ http://www.bartleby.com/66/38/43638.html ------------------------------------------- ----------------------------------------------------------------------- You are subscribed as slavitch () gmail com To manage your subscription, go to http://v2.listbox.com/member/?listname=ip Archives at: Archives: http://archives.listbox.com/247/Modify Your Subscription: http://v2.listbox.com/member/? & Unsubscribe: http://v2.listbox.com/unsubscribe/?id=3566138- aec10944-9j8zfoip
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- the end more on University dumps Cisco VoIP for open-source Asterisk David Farber (Jan 03)