[Asterisk-Users] why even use SIP

[Tom Samplonius]

On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith
<akohlsmith-asterisk at benshaw.com> wrote:
> On March 21, 2005 01:07 pm, Kevin P. Fleming wrote:
> > Well, let's see.. 99.99% of the available VOIP hardware only support
> > SIP, MGCP and H.323, but not IAX2. Is that a good reason?
> 
> No.  95% of the marketplaces uses Windows.  Drive the marketplace to use
> better protocols.

  That is rather subjective.  Many people consider SIP better than IAX since:

* There is a published standard, as opposed to some source code.
* It leverages a lot of existing standards already.
* It is media and addressing agnostic.  SIP can support any media, and
nearly any kind of addressing.  IAX is heavily mired in legacy
telphone standards.  Routing "3XX" to PBX A, and "4XX" to PBX B, is
80's PBX style telephony over IP.
* It separates media and signalling.  This is the biggest IAX problem:
 Why should a call switch have to get in the middle of the media to
make a call routing decision?

  I find that since Asterisk has an overly complex and still
incomplete SIP implementation, if the only exposure you have to SIP,
is via Asterisk, you should have a poor opinion of SIP.  Asterisk
makes SIP hard.  Asterisk doesn't even support a lot of the SIP
capabilities.  For instance, why can't the media type change during a
call-transfer?  Why does Asterisk have to be in the media path to
support a "t" or "T" type transfer if SIP INFO DTMF signalling is
used?  Why on earth does every two-bit RTP implementation support VAD,
but not Asterisk?



Tom