The VoIP Peering Puzzle�Part 38: SBC Architectures�SysMaster

SysMaster Corporation, headquartered in Walnut Creek, California, was founded in 1998 as a software development company, and has since morphed into a leading vendor of voice, video, and data communications equipment, serving both the traditional and next-generation telecommunications companies and service providers.

The company offers an extensive line of VoIP, IPTV, and wireless products and solutions that let providers build robust and scalable networks for delivery of next-generation services to subscribers. SysMaster’s customers include Calling Card Providers, Wholesale VoIP Providers, Hosted PBX Providers, Internet Telephony Service Providers (ITSPs), Competitive Local Exchange Carriers (CLECs), Incumbent Local Exchange Carriers (ILECs), Internet Service Providers (ISPs), Wireless Internet Service Providers (WISPs), IPTV Service Providers and others. Their product portfolio is very diverse, and includes VoIP billing server, gateway, and IP PBX systems, IPTV set top boxes, IP video phones, and Wi-Fi access nodes. SysMaster’s products and solutions are successfully deployed in over 60 countries worldwide.

For the session border controller market, SysMaster has developed the uniSwitch, which they designed to incorporate softswitch as well as SBC capabilities. The uniSwitch system provides SysMaster customers with a solution for secure and reliable peering between their own networks and the VoIP networks of their business partners. It offers a number of features to ensure connectivity with other VoIP systems and terminals, including the SysMaster line of VoIP gateways and billing platforms. It can also accept traffic from both registered and non-registered terminals, and can terminate traffic to gatekeepers, gateways, and softswitches.

The architecture of the uniSwitch supports both the SIP and H.323 protocols, with internal core engines providing softswitch, H.323 Gatekeeper, and SIP Registrar functions. The softswitch protocol conversion function can translate messages between SIP and H.323, thus bridging calls between VoIP systems that use these incompatible protocols.

The system also provides topology hiding, so that service providers can securely separate their VoIP network from other IP networks. With this capability, the system acts as a traffic proxy between out-of-network VoIP terminals and in-network equipment, thus preventing outsiders from seeing the true topology of the protected network.

One of the strong suits of the uniSwitch is that it offers very flexible routing of calls between VoIP networks. The product supports a number of routing methods, including Least Cost Routing (LCR), Average Success Rate (ASR) routing, Priority routing, Two-stage Routing, Preferred Routing, and Route Fail-Over.

The ASR option is interesting, in that the system monitors all the end terminals in real time, and automatically disconnects those whose ASR rates fall below critical values preset by the system administrator. With these multiple routing options, the service provider can select the most profitable and high quality routes for each call, thus increasing the call completion rates, and the associated call revenues. The system includes a mechanism to improve the network availability, by periodically conducting Layer 3, 4 or 7 remote service checks, and then re-routing calls to alternative remote terminals if the existing ones are not available.

The uniSwitch is designed to support four different modes of operation.

In Proxy Mode, both the signaling and the voice traffic flow through the softswitch. This is used for establishing connections to terminals behind a Network Address Translation (NAT) device, and for connections to terminals that need to remain anonymous. In this mode, the system exercises full call control, and can disconnect calls in progress if necessary.

With Routed Mode, the system performs direct control of signaling messages only, with the voice traffic exchanged on a gateway-to-gateway basis. With this mode, the system still exercises a substantial degree of call control, and can disconnect calls in progress if necessary, but requires less bandwidth overall.

There is also a Routed Mode w/o H.245 Support, where the H.245 call signaling is handled by the participating gateways, not by the system, which further reduces bandwidth requirements.

Finally, Static Mode allows both the signaling and voice traffic to be exchanged between the participating gateways, where the system consumes the least amount of bandwidth, but does not have control over calls in progress.

The system is offered in Standard and Carrier editions, to address the requirements of both emerging and established service providers. The Standard edition can support up to 500 concurrent calls, includes one Gigabit Ethernet controller, and is packaged in a 2U rack-mounted enclosure. The Carrier edition can handle up to 1,000 concurrent calls, includes two Gigabit Ethernet controllers, and is packaged in a 3U rack-mounted enclosure.

Further details on the SysMaster architecture and products can be found at Our next tutorial will continue our examination of vendors’ SBC architectures.

Copyright Acknowledgement: © 2007 DigiNet Corporation ®, All Rights Reserved

Author’s Biography
Mark A. Miller, P.E. is President of DigiNet Corporation®, a Denver-based consulting engineering firm. He is the author of many books on networking technologies, including Voice over IP Technologies, and Internet Technologies Handbook, both published by John Wiley & Sons.

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