Last time, we examined why SIP, or Session Initiation Protocol, is the kind of new thing that this industry ignores at its own risk.
This time, we’ll dive into how a SIP-based call is different than an NCS-based call — where “NCS” is shorthand for network-based call signaling, the method predominantly used by the industry’s PacketCable 1.0 specification. (That means it’s the spec most MSOs plan to deploy and most voice-over-Internet protocol telephony vendors are building their gear around.)
Before we even get started, know that both SIP and NCS take a gigantic leap from traditional telephony.
Oversimplified, that black phone bolted to the kitchen wall when you were a kid worked off the voltage of the actual phone line. VoIP phones work by sending a series of electronic messages that emulate how the old black phone worked.
The chief difference between SIP calls and NCS calls is this: One (NCS) gets real chatty with the center of the network, to figure out what to do. The other (SIP) doesn’t — its devices get chatty with each other to determine their to-do lists. They treat the network as a go-between.
Both work to call or receive calls from that old black phone.
As NCS is the method to be deployed by the MSOs doing VoIP, let’s start with that.
In the PacketCable world, you plug your phone into one of two devices. There’s an “S-MTA,” for “standalone multimedia terminal adapter,” or there’s an “E-MTA,” for “embedded multimedia terminal adapter.”
S-MTAs are like VoIP sidecars. They plug into the Ethernet jack of a cable modem. E-MTAs are both a cable modem and an MTA, in one box.
Economically, sidecar S-MTAs are cheaper than E-MTAs, but E-MTAs are cheaper than buying a cable modem and an S-MTA.
Without batteries, E-MTAs go for $70 to $80, technologists say. That’s about $30 to $40 in incremental costs compared with today’s volume-priced cable modems.
For this discussion, we’ll assume we’re dealing with an E-MTA.
When you turn it on, it gets two IP addresses: One for the cable modem, one for the voice activities of the MTA. The voice section also gets the electronic address of whatever softswitch is being used by that system.
Then, the chatting begins. The MTA pings: “Yoo hoo! I’m new here. Anybody there?”
The softswitch (which also goes by “call-management server,” or “CMS”) acknowledges: “I see you. Who are you?”
The MTA says what it knows about itself: “I’m a 2-line E-MTA.”
The softswitch gives it an identifying number and asks to be notified if anything happens. It’s specific about what could happen, like someone picking up the handset of the phone or dialing digits.
YOUR (NCS) CALL
So that’s you, picking up the phone. Say you’re calling your childhood home, where that old black phone is still bolted to the wall, and your parents still have the phone number they had when you lived there.
(That last part doesn’t matter to this discussion, except that in these times, it’s oddly nostalgic to know anyone who’s had the same phone number for more than 10 years.)
The softswitch collects the dialed digits, and places the call — either linking to the public switched-telephone network over a “media gateway,” or finding the destination E-MTA.
In the latter case, a similarly lengthy dialogue happens with the destination E-MTA, to receive the call.
When the call ends, the E-MTA tells the softswitch that you and Mom hung up.
Again, this is an oversimplification. Lots of other things happen throughout the call, especially in setting it up and tearing it down. The point is that the activities are tightly managed by the softswitch.
In fact, the E-MTA is too dumb to know much beyond how to notify the softswitch when anything happens.
In the world of SIP, just the opposite is true. SIP devices, which are known in the lingo as “user agents,” or “UAs,” treat their work like any other Internet communications program. Destination phone numbers become much like e-mail addresses, for example, that can be dialed from a handset, a PDA, a PC or a laptop. Functionally, UAs are like a combination of an E-MTA and a softswitch.
YOUR (SIP) CALL
Again, you decide to make a call. You could call Mom again, but the interesting stuff about SIP surfaces when you call another SIP device.
For one, you’re not solely dealing with a 12-digit phone keypad anymore. To reach Jane, you could click on her e-mail address. The SIP agent within your laptop will look up all the different ways she can be reached (or not).
Similarly, Jane — who might think she was a better candidate than you for the overseas trip — can decide whether or not she wants to take your call. She can also pick how she wants to take it – on her PC, office line, cell phone, and so on.
Logistically, here’s (loosely) what happens: The E-MTA inside your laptop (and Jane’s) is pre-registered with a “registrar” (potentially offered by your cable provider), which keeps a database of SIP device addresses.
Jane’s might be “SIP:email@example.com.”
Like a routing database, the registrar exists to know who you are, so it can find you when you get a call, or route you when you’re making a call.
Meanwhile, across the ocean, you click to dial. In the background, your SIP device is issuing an “invite” request to a location server, asking it to find Jane and invite her to the call.
Maybe Jane uses SIP on her PC and her phone. Your registrar passes your request to a location server, which doesn’t find Jane, so it passes the dialed digits to a redirect server, which locates her.
Back at the office, Jane sees your incoming call pop up on her computer screen. She answers — by donning a headset and clicking an icon on her PC.
In the background, Jane’s SIP device and your SIP device already negotiated an assortment of details about the call itself.
One example (of several) is which “codec” to use. “Codec” is tech-speak for “coder-decoder,” which is the thing that digitizes and squishes your voices for the ride over the wires.
Another example of the behind-the-scenes negotiations between your and Jane’s SIP devices is which port to use to establish the session for the two of you to talk. There’s back-and-forth chatting going on, but it’s happening directly between the two intelligent end points — not with the network.
The point: Contrary to NCS, SIP devices don’t need to ask the network what to do when you initiate a call, dial digits, talk and hang up. All they need is somebody to help route you to Jane.
There’s no easy way to say whether NCS or SIP is “better.” Aficionados of both techniques generalize it this way: NCS is sturdy, but not amenable to quick changes. It’s a little better at regulatory stuff, like enhanced 911 and CALEA (the Communications Assistance for Law Enforcement Act of 1994), the law-enforcement part.
NCS also contains specific methods of offering quality-of-service guarantees, which matter in terms of offering sustainably good quality. This will matter big-time, especially as video telephony enters the scene.
SIP is more agile and is riding a big innovation wave right now. Plus, it’s not just for voice — instant messaging and video telephony work with SIP, too. But SIP is potentially more prone to integration hassles. Already, a data-communications trade magazine ran a front-page story last month about SIP incompatibilities with the firewalls that reside in people’s home networks.
Note that early adopters are already making SIP calls, but they’re doing so via competitive providers like Vonage Holdings Corp. — and, shortly, AT&T Corp., Verizon Communications Inc. and others. It just seems too logical and economical to assume that SIP won’t spread like wildfire.
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