Archive for January, 2008

Vanilla, Chocolate, or Strawberry VoIP?

Tuesday, January 22nd, 2008

SIP comes in many flavors. This isn’t unusual because as you probably know, traditional Telephony comes in many flavors. The difference here is that TDM based telephony has had about century to evolve into relatively standard Vanilla, Chocolate, and Strawberry flavors.

VoIP is rapidly evolving so it’s possible to get Skypeberry, Googleroni, or Vonagenut, or any number of other exciting new flavors.

At Voxeo we speak SIP. Session Initiation Protocol is an IEFT (Internet Engineering Task Force) defined signaling protocol which is described in RFC 3261. The SIP RFC is a set of rules for how the protocol should operate and is open to interpretation by the party creating the SIP device or SIP code stack. So - how SIP functions can vary between device or software. We use a common set of rules in our SIP stack in order to play nicely with the rest of the SIP world and as such are compatible with most devices and providers that work with the G.711u codex.

However, sometimes we encounter inconsistencies in how our SIP works with different providers and devices. In fact, the primary purpose of the Voxeo lab is to test providers and devices for VoIP compatibility.

Voxeo Prophecy, as a SIP device (some like to call it an IVR - but I know it’s much more.. ;-) ) is able to accept SIP media in one of two ways, through SIP Registration or SIP Trunking. I’d like to take a moment to define these two transport mechanisms so that we can carry on future discussions with a common foundation. I believe this is needed because I regularly encounter different terms for the same things in VoIP - and unfortunately - it’s confusing, therefore - I’d like to clear the air and help define what we are doing at Voxeo with Prophecy.

SIP Registration is a way to establish and maintain a constant state-ful media transmission connection between a SIP Host (Server) and SIP device (Client). In SIP Registration, a SIP device will transmit a username and account password that is authenticated by the SIP Host. If the account and password is valid, the connection is accepted and calls may be passed between the two devices. If the authentication is rejected, calls will not be allowed. In SIP Registration, a constant state of communication is created and the pipe is monitored at regular intervals for availability.

SIP registration is normally used for one or several SIP devices - often SIP phones, that maintain a constant state of connectivity with an IP switch or SIP provider. Prophecy is able to utilize SIP registration to provide transport for small usage systems deployments, with usually less than 10 concurrent calls.

SIP TRunking uses IP Authentication between 2 devices, in order to establish authentication and allow transmission and receipt of data packets to occur. In effect, the IP Gateway of a SIP provider is known by the transmitting SIP device so that the media is allowed to pass. This method of transport requires the specific IP address knowledge of the SIP provider, and SIP end user device - so that a direct connection can occur between the 2 points.

Wholesale or large scale VoIP service providers and customers typically use this kind of VoIP transmission in order to allow large volumes of traffic to pass quickly through IP Gateways.

In most cases, small scale Prophecy deployments are able to effectively use SIP Registration services, while large scale deployments will utilize SIP Trunking scenarios for transport.

In future posts I will address both SIP Registration and SIP Trunking, as I discuss various topics around how Prophecy works with other VoIP Providers and Devices.

Until then, try out our BusinessVoIP service - which uses SIP Registration to maintain a state-ful connection to our Voxeo BusinessVoIP network. Just a teaser, there are many cool things that you can do with a state-ful connection to a VoIP Network. ;-)