[asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys

[Raj Jain]

> Somewhere in my mail archive I have a permission from Broadsoft or
> Cylantro to implement this - before they
> published it as a draft. They wanted Asterisk to include this
> functionality. After that, they published it openly
> as a draft, so there should be no problems.


It's probably also supported by the fact that this specification is publicly
available on BroadSoft's company website:
http://www.broadsoft.com/pdf/BW-SIPAccessSideExtensionsInterfaceSpec-R1301.pdf

If we speak about SLA specifically, they've basically extended SIP on three
fronts:

1. Additional parameters in the existing Call-Info header
2. Call-Info SIP event-package
3. Line-Seize SIP event-package

The Call-Info event-package pretty much accomplishes the same thing as the
standard dialog event-package (i.e. it notifies the state changes of the
shared line to SLA stations).

The Line-Seize event-package solves an arbitration issue that happens in
en-bloc signaling protocols such as SIP. It allows the entire SLA group to
be immediately informed as soon as one SLA station seizes the shared line.
This allows an SLA station to "win" the line. The other phones render the
line busy while one phone is dialing. I've seen different commercial PBX
vendors solve this problem in their own proprietary way.

There is one more interesting aspect to these SIP extensions -- they're
built on the concept of Call Appearance. Call Appearance means that a unique
line can have multiple calls active on it at the same time. So, you're not
only telling the phones which line underwent a state change, but also which
appearance of it.

Does the current Asterisk SLA implementation support the Call Appearance
concept?

--
Raj
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