[asterisk-dev] Naming of the new SIP Stack

[Matthew Jordan]

Hey everyone!

As you may recall, we had a discussion on this mailing list regarding
the name of the new SIP stack [1]. If I've combed through the mailing
list archive correctly, then we currently have the following totals for
the name of the channel driver (using the dial string as an identifier):

SIPNG: 1
GULP: 4
PJSIP: 9
SIP2: 1
SIP: 1 (with a rename of the old)

It's pretty clear that at this point, barring a *lot* of additional
commentary, PJSIP will win out. In the interest of keeping everything
moving forward, I think it's time we call it - and start referring to
the new SIP stack as "chan_pjsip".

First, thanks to everyone who participated in the discussion!

Second, if anyone would like to contribute some work to help us rename
all of this, that would be hugely appreciated. :-)

It isn't just chan_gulp that needs renaming/attention, but the SIP stack
in the res directory as well. Functions, CLI commands, and other items
need to be refactored. None of this is terribly hard, but it is
certainly time consuming, and some assistance would be greatly appreciated.

Thanks again!

Matt

[1] http://lists.digium.com/pipermail/asterisk-dev/2013-April/059669.html

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org