[asterisk-dev] SIP TCP/TLS, release policy and more (personal opinions included)
Kevin P. Fleming
kpfleming at digium.com
Sun Oct 19 23:15:49 CDT 2008
Johansson Olle E wrote:
> We do have a broken SIP channel in 1.6.0. UDP still works mostly as
> before, but the code for TCP/TLS support is just not going to work
> without a large amount of work. This was very clear at SIPit.
While I appreciate all the kind words in your email in my direction (and
I very much enjoyed working last week at SIPit 23 with you <G>), I will
once again ask you to stop stating that the 'SIP channel in 1.6.0 is
broken'.
As you have stated above, UDP support does not appear to have any
significant (or any, really) regressions from 1.4, and the TCP/TLS
support is clearly marked as 'experimental'. As such, I cannot
understand how publicly stating that the SIP channel driver is broken
does anyone any good, and it does great harm because many people will
read the first few lines of this mail (via RSS, blogs, etc.) without
understanding the context and Asterisk will gain a reputation as having
a broken SIP channel driver. This is very, very bad for the community
and Asterisk, and is unwarranted.
If your goal is to move Asterisk and the community forward to improve
the code and products/installations based on Asterisk, please focus on
providing constructive criticism (as most of your email is) and away
from making inflammatory statements (even when prefaced with a
disclaimer... as I noted above, many people will read these statements
out of context).
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
More information about the asterisk-dev
mailing list