First came the public switched telephone network (PSTN), then signaling system 7 (SS7), followed by session initiation protocol (SIP) and now SIP for the private branch exchange (sipX). One built on the other. This article covers what they are, so you can see what they mean to you when dealing with a customer who leases or owns a private branch exchange (PBX) or is considering a new PBX. Voice communications were first, and that led to communications for voice and data and Internet access.
PSTN is the telephone system based on copper wires that carries analog (a system of continuously changing values) voice. Subsequent telephone networks were based on digital technologies such as integrated services digital network (ISDN-)—the international communications standard for sending voice/data/video over telephone lines.
SS7 or C7
SS7 or C7 is a global standard for telecommunications defined by the International Telecommunication Union Telecommunication Standardization Sector (ITU-T). This standard defines the procedures and protocol by which network elements in the PSTN exchange information over a digital signaling network for wireless (cellular) and wireline call setup, routing and control. It is an international high-speed signaling backbone for the PSTN.
Note that signal switching points (SSPs) are telephone exchanges equipped with SS7 capable software. They are responsible for originating, terminating or switching calls. SSP can be an origination or destination point for signaling messages. Signal transfer points (STPs) are packet switches that receive and route signaling messages to their proper destination. Signaling control points (SCPs) are databases that provide information necessary for advanced call-processing capabilities.
SIP, which came out in 1996, is a sophisticated telecommunications protocol that provides out-of-band signaling and a data interface between phone company switches for the express purpose of reducing congestion in the PSTN. It is also the Internet protocol (IP) telephony signaling protocol or standard that enables/controls the connection, communication and data transfer between two computing endpoints (e.g., telephones) for initiating, modifying and terminating interactive user sessions involving multimedia elements. In other words, it is the “control” mechanism that is used by voice and data networks.
SIP is text-based, similar to HTTP and SMTP, for managing the handshake procedures for beginning and ending real-time communications between IP network end points/devices. It is used to enable human-to-human communications that might include voice, video, chat, interactive games and virtual reality.
SIP protocols vary greatly in their purpose, but most specify:
- Detection whether the network is wired or wireless and what the endpoint hardware is
- Negotiation of various connection characteristics
- How to start and end a message
- How to format a message
- Error correction
- Termination of a session or connection.
SIP protocols development began with PSTN for call processing functions and was implemented in the phones at the endpoints, then it advanced to SS7 systems that were centralized and had advanced call-processing features. SS7 carries a description of the media content (packet streams) for the real-time transport protocol (RTP) that carries the actual voice or video content. In reference to voice over Internet protocol (VoIP), it routes the calls, authenticates, and authorizes users for services. Today there are more sophisticated signaling methods that have come out of SIP, one of which is sipX.
SipX began in 1999 and has been touted as a next-generation IP-PBX solution. An IP-PBX is a private branch exchange (telephone switching system) that can switch calls between VoIP users on local lines, VoIP users and telephone users, or between two generic users. Now, using an IP-PBX, you can converge the past separate networks for voice and data and provide a single line for a user to access the Internet, VoIP communications, and traditional telephone services.
SipX is an open or nonproprietary software implementation of SIP—meaning any developer can download it from the SIPfoundry Web site (www.sipfoundry.org) for free and program a Linux-based PBX or “soft” PBX. This SIP-based communications system can be used to replace a PBX. SipX can:
- Allow attached phones to make calls to one another and connect to other telephone services, including the PSTN
- Be used for toll bypass
- Be used as a VoIP solution, with voicemail and auto attendant.
- Run on a standard Linux operating system, with other operating system development work underway
- Work with any SIP-compliant gateway, phone or application.
SipX was originally introduced by the company Pingtel. Later came the SIPfoundry, a nonprofit organization dedicated to the development and adoption of open source SIP-based communication solutions. SIPfoundry developed the entire commercial code base open source project under sipX.
SIPfoundry’s projects focus on all aspects of IP communications for the user, the agent, the enterprise or the carrier. They also provide interoperability testing solutions.
Pingtel, Woburn, Mass., offers SIPxchange, the commercial version of sipX that uses the same code, but is more hardened. New businesses can use Pingtel’s SIPxchange to deliver VoIP instead of extending the time division multiplexing (TDM) technology of transmitting multiple signals simultaneously over one transmission path. TDM enabled the telephone companies to migrate from analog to digital on all their long-distance trunks.
There is a downside to IP-PBX conversion: high cost. However, there are many payoffs. First, the developing software is free, interoperable, scalable and delivers cost savings, bringing users to the cutting-edge of communications applications.
SipX and VoIP
Voice and data networks are converging, and IP telephone and VoIP are driving this convergence. Predictions allude to gradual, yet substantial, growth in businesses with the majority of that growth being for mid- and large-sized businesses.
The sipX open-source software solution can help a startup whose PBX or IP-PBX is based on SIP. SipX becomes a software-based PBX that runs on Linux, does VoIP and is interoperable with a variety of standards-based telephony equipment.
How sipX can benefit customers
Contractors will be interested to know that the demand for sipX is coming from the customer. Users have found that it doesn’t make economic sense to upgrade their infrastructure to deliver voice services over the PSTN when it is easier to use the data network to deliver voice. They already know standard SIP, and some people want a product that will work with that protocol. SipX does this, plus a lot more. Some advantages to a sipX communication system are as follows:
- The system is nonproprietary and can use different manufacturer’s products for the hardware. The customer would have more flexibility because they could pick hardware to use with it, thus saving money.
- The sipX solution is very easy to use because its management system runs on an Ethernet network. The hardware is like a “node” on the network.
- This works for small businesses as well as large. It is uncomplicated to go from a small business (using a $500 server) to a large enterprise (requiring several servers).
- Moves, adds and changes are easier because that administration is now integrated within the IT environment.
- This software now also becomes a Web service to employees because they can get PBX information right on their PC or laptop.
The architecture of sipX is its best feature. It uses individual servers to handle different activities such as configuration, registration, authentication, status, parking and conferencing. The architecture is modular and consists of three main building blocks/servers: the sipX communications server, the sipX media server and the sipX configuration server.
And, while sipX packages these components to function as a SIP PBX, each server can stand alone. SipX is user-friendly with screens providing easy access to assigning auto- attendants, to handle a dial plan and call routing, to scale up or down, and for voicemail use. The user or administrator can define features, such as call forwarding, call hold and retrieve, call park, call waiting, calling line identification, calling party name identification, and conferencing. Visit www.sipfoundry.org to see the brand name phones supported as well as gateways and servers.
Martin J. Steinmann, founder and member of the board at SIPfoundry, said: “There are a few things about sipX really important to contractors. SipX is a second-generation VoIP system for the enterprise entirely based on the SIP standard and built with ease of use as the prime objective. As such, sipX is a solid system that offers all the features customers demand, and it is fully competitive with legacy PBX solutions.
“SipX is being successfully installed both at small enterprises as well as large enterprises and is growing rapidly. Significant economic advantages, both for contractors and end customers, result from the fact that standard server hardware is used and that any SIP compatible phone and gateway can be chosen.
“No longer are customers or contractors locked into a specific vendor’s solution. Therefore, a sipX solution is typically 40 to 50 percent less expensive while enabling true real-time communications far exceeding voice-only requirements. Through Pingtel, 7-by-24 support, training, documentation and everything else you would expect, is available to contractors and VARs.”
A customer may want to upgrade to converge their voice and data services. If you can recommend software and hardware (for example, sipX) that can help them now as well as when they grow, you have given them an economic benefit. EC
MICHELSON, president of Jackson, Calif.-based Business Communication Services and publisher of the BCS Reports, is an expert in TIA/EIA performance standards. Contact her at www.bcsreports.com or firstname.lastname@example.org.