SIP Trunking


What is SIP Trunking?


SIP, or Session Initiation Protocol, is the standard communications protocol for voice and video in a Unified Communications (UC) solution across a data network. A SIP trunk replaces the need for traditional analog, T1-based Public Switched Telephone Network (PSTN) connections with termination instead provided over a company’s public or private Internet connection through a SIP provider. These SIP providers, often referred to as Internet Telephony Service Providers (ITSP), provide PSTN service on a per minute or channelized pricing model.


The per-minute pricing model is fairly self-explanatory, with a set rate per minute of usage. A channelized pricing model typically provides nearly unlimited minutes on a set number of channels, or call paths. For example, a company can purchase 10 channels and make use of unlimited minutes on those channels, but can only have 10 simultaneous calls.


Many companies already use VoIP within their PBX on the Local Area Network (LAN) to connect to IP phones. SIP Trunking also uses VoIP to take advantage of shared lines, such as a company’s Internet connection, to allow more flexibility in communications. Traditional legacy systems, that aren’t already VoIP-capable, can be connected using common VoIP gateways to take advantage of SIP trunking and reap the significant cost benefits.


The Benefits of SIP Trunking


The cost savings and communications benefits of SIP trunking are substantial. Your company is most likely experiencing high costs with an existing PBX, while still having the constraints of the limited communications technology provided by your current Telco. High costs may be incurred through a combination of monthly phone bills, which include charges for incoming phone lines, long distance charges and IT and maintenance fees, all of which can be drastically reduced or eliminated by a SIP Trunking provider.


SIP Trunking allows companies to only pay for the number of lines they need as opposed to getting locked in to excess analog lines or partially-used T1s and PRIs. The savings are realized either by purchasing only the necessary number of channels, or by paying only for minutes used. This allows companies to make more efficient use of communications costs and reduce or eradicate wasted resources.


SIP Trunking eliminates the physical connection to a phone company. There are no hardware, wiring, or circuit boxes to maintain for connection to the PSTN. Reducing multiple phone lines into a single point of entry drastically reduces charges for incoming lines and the IT cost associated with the maintenance of those lines. (Some organizations prefer to maintain standard lines for faxes and alarms.) A phone number, or Direct Inward Dialing number (DID), is less expensive when purchased with a SIP Trunk. Traditionally, when a DID is obtained from a phone company, charges are applied for the DID, IT and maintenance services, and the hardware connecting the shared physical lines or channels. A DID provided without these infrastructure costs is more affordable.


SIP Trunking with VoIP increases reliability of services by providing a level of redundancy. When system failures and emergencies occur, SIP Trunking providers can reroute services to a redundant data line or forward the PBX to mobile phones to keep your business up and running.


Hardware Requirements


Most Ericsson-LG telephone systems come equipped with a basic capacity to run SIP trunks (The exception is the UCP2400 and LIK1200 which require a VoIM card) without the need for further hardware or licenses. The iPECS range also do not require any additional 3rd party hardware to facilitate SIP trunks (e.g. Ingate SIPerator), all of our systems work via NAT using your internet connection. To expand the number of SIP trunks on the iPECS is simply a case of either adding additional VoIM hardware or VoIM licenses.




The Ericsson-LG eMG80 / eMG800 / UCP / LIK have all been approved for use with a large number of SIP carriers around the world including ThinkTel (Canada), Telstra (Australia), Skype Connect (Global) and Broadsoft (Global) to name but a few. At Nine-One-One we will provide your dealer with all the nessisary documentation and technical support to setup and configure the SIP trunks for any approved carrier.


Looking To Test ?


At Nine-One-One we have a testing and development SIP Proxy server which will help you to ensure that your network is ready and able to work with SIP trunks. Our dealers can come on-site and test both the inbound and outbound functionality BEFORE you have to commit to a contact with a carrier. This eliminates the possibility of having to resolve expensive network or firewall issues AFTER you have migrated.


Try it out ..


If you would like to see how the voice quality compares to your traditional CO / PRI service or to simply see how it works, download the MicroSIP client HERE, unzip the file, run MicroSIP.exe and DOUBLE CLICK one of the contacts.




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