> Dome Charoenyost wrote:
>
> > Can someone explain about SIP,SIP-T and SIP-T ?
> > and Asterisk support SIP-I,SIP-T or not ?
>
> It is a way to map and/or encapsulate some values particular to PSTN
> signaling protocols (such as ISUP) back into SIP so that SIP can serve
> as a minimally viable transport between two nodes that speak those
> protocols, i.e.
>
> ISUP ----> SIP/ISUP gateway ----> IP ----> SIP/ISUP gateway ---> ISUP
>
> Asterisk does not really support these because it is a back-to-back user
> agent, so it does not need to follow RFC rules regarding conservation of
> special headers or data elements in a transaction the way that a proxy
> does. Asterisk can and will regenerate new call legs scoped to the
> interface technologies it actually supports and cares about.
>
Supporting them has nothing to do with being a B2BUA, PBX or softswitch.
They are variants of SIP, and as such they would require a proper
protocol stack to support. More at the end on this.
It should however support the RFC rules on at least a single channel,
which it currently violates some of them, and at least some are known
deficiencies. Quick and simple example, TCP is not optional per the
RFC, both UDP and TCP for transport are required. There are a few other
examples like that, where required non-optional things are missing and
afaik they are all known. And per the trademark TOS if its "asterisk"
its unmodified software from digium, so forks and such that do support
this are not supposed to be "asterisk".
However, to expand on what you said, it can support SIP-I (the isup
thing in your example) and translate the data from the ISUP channel to
the SIP-I, but that would require a protocol stack to be written or
found, or the existing one modified. It would also require a way to
toggle the I/T stuff on/off such that you dont try to speak SIP-x to
something that does not understand it (it can cause confusion on the
other end even if all the SIP only headers are correct and present).
Further you can use SIP-I/SIP-T anywhere you can use SIP (providing it
doesnt confuse the other end) its a protocol stack like SIP. Just as
you can make a SIP call originated from asterisk, with or without
another leg being bridged in, you would be able to with SIP-T. Think of
it more like chan_sip-t in the asterisk design model, where its a
protocol stack like anything else, the fact that its mostly the same as
SIP shouldnt matter too much.
--
Trixter http://www.0xdecafbad.com
Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!
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