SIP Trunking: Part 1

By Adam Friedmann

You may have heard the term SIP Trunk being thrown around and wondered what all of the excitement is about. Well in order to understand what SIP Trunking is we should probably understand a little bit about how Voice over IP (VoIP) works. Session Initiation Protocol or SIP is a protocol which sets up and tears down a call that is carried over an IP network. Many organizations use SIP as a means to setup calls within their VoIP infrastructure. If I want to place a call to a colleague via our VoIP network, SIP will be used to setup the call and Real-time Transport Protocol (RTP) will be used to transmit my voice over our network.

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If I decide to place a call to my wife to let her know I will be late from work, SIP will still be used to setup the call but now we must have a way to interface with the telephone company to send the call to my home phone number. A gateway serves this purpose. The gateway will take the IP packets that my voice is translated into and convert it to analog information that can be sent over the PSTN. Likewise, my wife’s voice will be translated back into IP packets by the gateway so that they can be transmitted across the network.

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One of the reasons that an organization purchases a VoIP infrastructure is to save money on toll charges. Let’s use the example of making a call from my office in Tucson to a customer in Germany. If I have an office in Germany that is linked via a Wide Area Network (WAN) to my office in Tucson I can add a gateway in Germany and route my calls across my network to that gateway. The gateway would place the call on the local PSTN in Germany bypassing toll charges. If our organization had offices all over the world with WAN links I could place a gateway in each office and bypass toll charges when calling customers or colleagues local to those areas.

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Smaller companies might not have the staff to maintain this sort of infrastructure. This is where SIP Trunks enter our story. SIP Trunks are offered by Internet Telephony Service Providers (ITSP) and allow an organization to see many of the benefits of a VoIP network without having to make a large capital outlay in the purchase of network infrastructure. The gateways that would be maintained in a standard VoIP implementation now move into the data cloud and are maintained by the ITSP. The organization purchases a SIP trunk from an ITSP and integrates it with an internal IP PBX. Connectivity is usually handled via a session border element that is maintained by the provider. Now when I make a call to my wife to let her know I am going to be late, the call is handed off from my IP PBX to the SIP trunk via the session border element and the provider routes the call to my home number via the PSTN. The organization only has to maintain the IP PBX.

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We’ll get an opportunity to learn more about SIP Trunking and how it relates to the OpenText Fax Server and OpenText Fax Gateways in a future blog. If you are interested in learning more about SIP trunks now you can visit these sites:

The OpenText Fax Server works with Global Crossing and Broadvox sip trunks.


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    In part 2 of the 3-part SIP Trunking series, we go beyond the simple implementation approach of just providing an IP address and explain the additional work required to get any fax server to interoperate with the SIP Trunk.
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