Re: [rtcweb] Bridged line appearance? (Re: Usecase & architecture: Browser application with separate webserver & voipserver)

[Matthew Kaufman <[e-mail]>]

On 9/7/2011 4:10 AM, Harald Alvestrand wrote:
>
> Since more people than I may be confused, I'm forking a subthread...
>
> what is a bridged line appearance, and why is it hard to do in SIP?
>
> My terminology cache is totally blank.
>

Key system (and home phone) emulation requires two things that are hard 
to do (because there's no final specification, not because it is 
impossible) (and because there's 3+ alternatives, and almost no phones 
implement more than one of them) in SIP:

1. Shared line appearance

This is where you can be on a call on one handset, see that the line is 
in use at the other handset. Place the call on hold at the first, pick 
it up at the second.

In business PBX cases, this is used for executive/assistant cases. In 
key system emulation it is used for "Bob, pick up the call holding on 
line 3". And for home phones it is the usual "put the phone on hold in 
the kitchen, pick it up in the den." (My wife wanted this functionality, 
so I had to add proprietary support for shared line appearance on my 
home PBX.)

2. Bridged line appearance

This is where you can be on a call on one handset, see that line is in 
use at the other handset, and pick up the second handset to join the call.

In the business PBX and key system case this is sometimes used for 
supervisors to join a call to help, but the real common case is the home 
phones... "kids, go pick up the extension in the living room and talk to 
grandma."

With POTS, this works by simply paralleling two handsets on the same 
copper pair. For SIP it requires everything shared line appearance does 
*plus* automatic barge-in + conference on pickup.

The BLISS WG has been working, for a long time, on taking the various 
interim proposals and creating a standard out of them, but we're still 
not there... and yet without it, there's no way to use phones from two 
different vendors to accomplish this.

Whereas with WebRTC implemented *without SIP*, it is fairly easy to 
build a web app that runs on multiple browsers from multiple vendors and 
implements this.

So this type of ability to innovate and implement applications without 
going through the standards process is why we *definitely* don't want 
SIP baked in to the browser.

Not to be confused with the arguments against SDP offer/answer, which is 
only slightly affected by the above use cases (in that you can do 
smarter things at the bridge for bridged line appearance if you aren't 
doing a reinvite/re-offer/re-answer but instead have on hand all the 
capabilities of the first device when the second goes to join).

Matthew Kaufman