[VoiceOps] Phone auth for incoming calls?

Alex Balashov abalashov at evaristesys.com
Wed Aug 8 15:36:40 EDT 2018


On Wed, Aug 08, 2018 at 12:21:09PM -0700, Carlos Alvarez wrote:

> So...who else on the list uses TCP and has any comments about it?

We are not an ITSP and are Polycom-only with a trivial number of
endpoints, but we do use it and it works just fine. 

However, we have numerous customers, some of whom use TCP predominantly
for thousands of endpoints. It works just fine.

In terms of downsides:

In addition to a historical lack of (RFC 3261-mandated) support, there
are of course theoretical trade-offs involved in using TCP. There's
more overhead, and connection state to be maintained on the server side,
which of course consumes resources — resources considered trivial
nowadays, but once upon a time, when RFC 3261 was ratified (2002),
perhaps not. As with all things TCP, it can also present a DoS vector if
you don't limit the number of connections somewhere. 

The congestion control/end-to-end delay aspects of TCP are certainly not
as important now as they were at a time when the public IP backbone and
was in an entirely different place in its evolution. Also, nowadays the
congestion/windowing algorithms used in TCP can be tweaked to something
more efficient.

I think the most damning thing about using TCP is perceived to be the
relative difficulty of failing over TCP session state to a different
host. UDP does not require connection state, so as long as you have some
means of handling requests in a relatively stateless fashion, things can
just carry on as they did before in the event of an IP takeover without
anyone having to "reconnect". This is one area where the big enterprise
boxes certainly trump the open-source ecosystem, where transparent TCP
failover *for SIP* doesn't really exist, although in my opinion the
whole issue is getting a bit moot with the way cloud infrastructure and
virtualisation networking is evolving.

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/


More information about the VoiceOps mailing list