Super Charging the SIP Trunk

I was on a panel about Supercharging the SIP Trunk Sales by Broadvox. It sounded similar to the panel I did at CVX West 2010 on Upselling the SIP Trunk sale.
This one had to go a little more basic to What is SIP? SIP Trunking is how carriers power dial-tone to an IP-PBX. Beyond the definition of the term, there is the concept that this specification for a voice packet to provide dial-tone is the foundation of the Next-Gen Communication platform. Start with the SIP trunk as the basic circuit needed for communications. Then other applications that utilize SIP sessions can be integrated into the platform. (In many cases, the platform is the PBX or the softswitch.)
One point that I made: All voice will soon be Voice over IP. Cellular is. Long Distance is. Dynamic T1 is. Cableco digital voice is. AT&T is turning off the PSTN with the approval of the FCC in less than 10 years. SIP has become the de facto protocol for Voice over IP - beating H.323, MGCP and others. Agents (and customers) have to get on-board with this fact. SIP is here to stay.
Yes there are issues. Most notably Fax-over-IP (which many companies are working on, T.38 not withstanding); HD Voice; alarms, elevators, credit card processing machines; and inter-operability between the SIP provider and the gear attached to the Trunk. Since SIP is a specification of about 30 RFC's, it is interpreted differently by manufacturers and providers. Hence, inter-operability is extremely important. It is not like a PRI, which is a standard, with just 2 configurations available in any class 4 or 5 switch or any PBX system. That makes inter-op easy. Today. Maybe not so much in 1988.
Our panel then went a little crazy talking about Unified Communication and all the possible UC components that could be mounted on that SIP trunk - like SMS/text, IM/chat, Video, Presence, conferencing, ACD, IVR, etc. Unfortunately, it's not that easy and most of the integration is with the gear (PBX, IAD, softswitch, SBC).
During Q&A, some asked how to explain SIP to a customer. I said, Don't. Would you explain how the engine works in a car? Focus on the benefits, the reliability, the way it will help the business and you won't have to worry about saying VoIP or SIP.
The panel had to explain that SIP trunks mean different things to different carriers. In some cases, a SIP trunk is just a call path. In other cases, a SIP Trunk is a circuit containing many call paths. It's confusing for the agent and the customer.
There are two ways to deliver SIP Trunking: over an IP circuit and over the Internet. Big difference. SIP sessions over the Internet lack quality of service. Jitter and latency will affect the voice quality. SIP Trunks that ride private line, virtual circuit or MPLS circuit have not only the best call quality but enhanced security.
Many ITSP's started out as ISP's (or at least with an ISP infrastructure). At the NOC, the ITSP has aggregation circuits for DSL, MPLS, private line from ILEC's and CLEC's. Some even have aggregation with a cableco. This means that the ITSP - in its network footprint - is delivering VoIP service to you on a private network with QOS of some level. The customer is receiving Internet over the same circuit but the egress to The Web is AFTER the aggregation point and separate from the softswitch.
Major MSO's deliver their Enterprise VoIP product as a physically separate VLAN. That offers QOS and security as well.
At XO,  Pete Davis and I have collaborated on a concept called The UC Sandbox. XO has all the components for UC available for sale. Hosted MS Exchange, Blackberry server, Hosting, Collocation, IM/chat, Hosted PBX, SIP Trunking, Contact center application, web & audio conferencing, video conferencing, data storage, Anywhere service, and more can be provisioned for the customer as a single individual service or as a bundle of components. The bonus for VAR's is that if they have a business selling email, security, what-have-you, they can mix-and-match components from XO or from a variety of providers to put the best solution together for the customer. That is the true value of VoIP, SIP and the future of Cloud.

SIP Trunking

SIP: Understanding the Session Initiation Protocol (Artech House Telecommunications)

Western European SIP Trunking Market, 2007-2012

How to Edit a Sip Trunking

Voice over Internet Phone service providers use Session Initiation Protocol, or SIP, to provide customers with access to the private broadband telephone network. SIP includes server information as well as address, username and password and other information needed to access VoIP networks. SIP information stored in hardware or software IP phones ensures that calls route through the correct network before connecting to external VoIP phones or landline telephone systems. If you switch VoIP services or providers, you must edit the SIP settings before you can use your phones to place or receive calls on the new network.

Instructions

Reconfigure Software SIP Phones

1) Launch the software VoIP phone application on your computer.

2) Click the "File" or "Tools" menu or tool bar link. Click the "Preferences" or "Settings" menu option to enter the software phone configuration or settings utility.

3) Enter the new username and password assigned to you by the new VoIP telephone service provider. Replace the entry in the "External Server," "VoIP Server IP Address" or "SIP Server" field with the public IP address of the new provider's SIP site IP address. Save the configuration settings and exit the application.

4) Restart the software phone program and wait for the application to connect and authenticate with your new VoIP provider's SIP server. After the software establishes the connection, use the software VoIP phone to place and receive calls via the provider's server on the Internet.

Change SIP Settings on a Hardware IP Phone

1) Unplug the AC power adapter from the hardware IP phone and leave it disconnected for about 30 seconds. Reconnect the AC adapter and power on the IP telephone. Wait for the hardware IP phone to initialize and for the configuration IP address to appear on the phone's display screen. The IP address should be similar to "192.168.1.101."

2) Go to a computer connected to the same router as the hardware IP phone. Open a web browser application and browse to the IP address displayed on the phone. Enter the username and password and log in to the IP phone's web-based configuration utility.

3) Click the "IP Settings," "SIP Settings" or "VoIP Account Settings" link in the configuration utility. Replace the entries for the VoIP account username and password, SIP server name and SIP server IP address with those provided to you by the new VoIP service provider. Click "OK" or "Update."

4) Wait while the configuration utility downloads the account changes to the hardware IP phone. Reboot the IP phone when prompted to do so.

5) Wait for the IP phone to boot and connect to the new VoIP network. Use the hardware IP phone to place and receive calls using the new VoIP provider's network.

Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation 

Western European SIP Trunking Market, 2007-2012 

U.S. SIP Trunking 2008-2012 Forecast: SIP Trunking Wins This Battle, But Who Wins the War?