2007-10-22 - Open-Source PBX Face-Off: SIPxchange Vs. Asterisk
Businesses pondering the deployment of an open-source IP PBX face two fundamental choices: Digium Inc.'s Asterisk and all its varieties and Pingtel Corp.'s SIPxchange ECS (Enterprise Communications System). Both technologies have the same basic goal: to serve as a software-based PBX solution that works well and can scale efficiently and cost-effectively, from small to large installations.Asterisk: The technology's goal is to serve as an IP PBX based on protocol internetworking standards, such as H.323, PSTN (public switched telephone network), Skinny, MGCP (Media Gateway Control Protocol), IAX (Inter-Asterisk eXchange) and SIP (Session Initiation Protocol), enabling it to interconnect with a variety of different systems. (For more information on VoIP standards, see VoIP-News's IP PBX FAQ.) While Asterisk converts all major signaling protocols, it is not a SIP proxy server that provides global routing of SIP sessions.
SIPxchange ECS: The technology's core infrastructure is rooted in SIP IETF's (Internet Engineering Task Force) standard for multimedia conferencing over IP. No media streams pass through the SIPxchange server. Instead, streams are passed directly between endpoints with just call-control messages relayed to the server. This approach places reduced workload requirements on the IP PBX system, giving the SIPxchange server an architectural advantage.
Asterisk: Asterisk offers both classic PBX features and advanced features. The IP PBX interoperates with traditional standards-based telephony systems and VoIP systems. Asterisk offers numerous sophisticated features that are usually associated with large, high-end (and high-cost) proprietary PBXes, such as roaming extensions, on-hold and on-transfer music, and dial by name.
SIPxchange ECS: SIPxchange ECS has a feature set that closely parallels that of Asterisk — including the sophisticated, high-level features — proving that basing a system completely around standard SIP can be used to create a feature-rich IP PBX.
Asterisk: Voice quality is hampered in Asterisk by its traditional PBX structure. Bandwidth is wasted using lines to feed voice and signaling data into the IP PBX. This makes the system vulnerable to voice jitter and delay.
SIPxchange ECS: Unlike Asterisk, SIPxchange ECS doesn't route media through the server. Instead, peer-to-peer media routing is used to provide enhanced voice quality with reduced delay and jitter.
Asterisk: Easy administration is not an Asterisk hallmark; the primary administration tool is a Linux command-line interface. System configuration is handled via an assortment of text-based configuration files. However, third-party configuration tools, such as AMP (Asterisk Management Portal), are available.
SIPxchange ECS: The SIPxchange ECS administrative Web interface allows for fast and easy system setup and management. New users can be provisioned and configured separately or as a group. Since SIPxchange ECS is entirely SIP-based, adding gateways and other SIP devices is as easy as adding a user to the system.
John Edwards on October 22, 2007
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