Providing SIP Trunks

“With the continued accelerated growth in VoIP technologies carriers and ISPs are now actively offering clients SIP Trunks as a lower cost alternative to traditional PSTN interconnect

What is a SIP Trunk?

In a traditional PSTN interconnect a “trunk” is typically defined as an E1/T1 interconnect which provides 30/24 channels of voice/data to a client. A SIP Trunk is a collection/ grouping of VoIP channels/ports made available to a client to interconnect voice/data.

SIP Trunk benefits :-

  • Reduced cost
  • Security of service via VPN
  • Guaranteed IP Bandwidth
  • Billing
  • Intelligent routing
  • DDI or DID (Direct Inward Dialling) support

Unlike a PSTN interconnect a SIP Trunk can consist of 1 or more SIP channels/ports – the actual number made available depends on the service offering and the end clients bandwidth requirements. A SIP trunk provides the guaranteed IP bandwidth needed to deliver the number of channels/ports offered over the selected VoIP codec.

In comparison to fixed ISDN line, the main driver is a significant reduction of telephony costs to the end customer and the ability to provide highly managed services such as intelligent call routing, billing and security of service through a VPN connection.

SIP Trunk Configuration 1 - PSTN over SS7/ISDN

There are a number of network topologies that allow carrier/ISPs to offer SIP Trunks. The first to consider is deploying a PSTN Media Gateway that connects to PSTN over SS7/ISDN and offers the SIP Trunking directly to the end client over a VPN. In order for an end client VoIP PBX to communicate to the PSTN Media Gateway, it needs to ‘register’ with the Media Gateway over SIP.

PSTN media gatewayThe SVI_MG provides full support for the registration process, which once completed then allows uni-directional calls between the PSTN and clients VoIP PBX/end point.

As part of this process the gateway will store the relevant DDI/DID (Direct Inward Dialling) against a registered SIP device to allow routing to the client’s VoIP equipment through the PSTN.

Carriers/ISPs need to provide a billing platform to invoice clients for calls, either integrating the PSTN Media Gateway into their existing billing platform via CDR’s, RADIUS or SQL interfaces provided on the SVI_MG or by purchasing the fully integrated Wholesale Billing package.

This configuration also allows carriers/ISPs to offer a more intelligent value added service by offering Least Cost Routing (LCR) enabling the routing of client calls to the most cost effective end carrier. Again this service can be offered by integrating the SVI_MG into the clients LCR engine over RADIUS / SQL or by using the LCR engine offered on the SVI_MG.

SIP TRUNK Configuration 1

SIP Trunk Configuration 2 - Utilising a Session Board Controller

SIP TRUNK Configuration 2

An alternative configuration is to deploy the PSTN Media Gateway in conjunction with a Session Board Controller (SBC) as shown. This allows carriers/ISPs to utilise incumbent equipment, using the SBC to offer the VPN connection to the end client with the SIP Registration process and SIP Signalling occurring directly between the SS7 Media Gateway and the Session Board Controller.

PSTN media gatewayAs with configuration 1 the carrier/ISP can also add a Billing Platform and Least Cost Routing (LCR) functionality either by integrating over RADIUS, SQL, CDRs or using the built in LCR and billing engines on the SVI_MG.

SIP Trunk Configuration 3 - Pure VoIP Environment

Class 4 SoftSwitchThe third configuration to consider is the provision of SIP Trunks in a pure VoIP environment. As shown in the network diagram below the deployment of a Class 4 SoftSwitch allows carriers/ISPs to offer SIP Trunks. The required SIP signalling is done via the Class 4 SoftSwitch with the RTP/RTCP Media transmitted directly between the 2 end points.

With the addition of a Least Cost Routing Engine (3rd party or integrated) this allows carriers/ISPs to offer the value added service of low cost national or international calls.

As with configuration 1 and 2 the carrier/ISP will need to consider whether to use an integrated or 3rd party Billing platform.

SIP TRUNK Configuration 2

SIP Trunk Products

Squire Technologies has deployed SS7 to VoIP PSTN gateways worldwide and assisted many clients to achieve PSTN interconnect, providing remote or on-site installation and commissioning of products quickly and cost effectively.

Coupled with Squire Technologies post sales support and product enhancement services, allows Squire

Technologies to deliver SS7 based signalling products.

Contact UsFor further information on how Squire Technologies can provide an effective SIP Trunk solution click here or download the presentation below.

PDFMore Case Studies

SVI MG Media Gateway

To connect over SIP or H.323 there are 3 models 1000, 2000 and 8000. Click model configurations below for details.

This entry level product is a fully featured SS7 to VoIP signalling solution, scaling from 1 E1s to 72 E1sMid range solution with support for STM-1 and DS3, scaling from 8 E1s to 224 E1sA high performance blade server, supporting STM-1 and DS3 interfaces. Scaling from 16 E1s to 1000 E1/T1s

Class 4 SoftSwitch

Class 4 SoftSwitch enabling the delivery of reliable VoIP traffic and services over multiple IP networks.

Boxed ProductRemote Install

Case Studies


Download PDFPresentation featuring SIP Trunks, Low Cost SS7, Innovative SS7 VoIP solutions and download SIP TRUNK presentationGSM Deployments typically used by carriers and ISPs.

SIP Trunk featured products

SS7 Media Gateway - SVI_MG

VoIP Softswitch - SVI_C4


Multi Vendor Solution

SVI_MGC SS7 Media Gateway Controller

Are you looking to interconnect to an SS7 network with existing VoIP Gateway Infrastructure from any of the following major vendors ?

Cisco SS7 over IP
Alcatel-Lucent SS7 over IP
Audiocodes SS7 over IP

Cisco

AS53xx, AS54xx, AS58xx & MGX8880 …more +

Alcatel-Lucent

Max Lucent TNT, Lucent APX …more +

AudioCodes

Mediant 1000, 2000, 3000, 5000, 8000
Orca GX Series …more +

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