Trunk Analysis - SIP Bandwidth Management
The Need for a New Communications Architecture
Many enterprises are using traditional communications architectures – a wide variety of individual PBX’s and key systems connected individually to the PSTN by a combination of T1 and POTS. As time moves forward, many organizations are looking to move to a new architecture centered on SIP to take advantage of technology to yield reduced costs and increased capabilities.
SIP – New Challenges in Traffic Engineering
In the past, well worn methods of traffic engineering were used to plan voice networks. Calculations such as Erlang B and Erlang C ruled the day. Since T1’s were used, companies typically added circuits in blocks of 24. SIP changes all of this. The key metric of SIP traffic engineering is the number of concurrent calls needed. This not only equates to the way the service providers bill for PSTN SIP circuits, it relates directly to bandwidth required. To properly engineer the network at each phase of the deployment, you need to calculate the number of concurrent voice channels needed. These calculations must be made for both on-net and PSTN calling. Many companies simplify this by guessing. For example, if a location had four POTS lines, they simply assume they will need four SIP circuits from their service provider. This can be a costly mistake. It does not take into account:
- How many actual concurrent calls are required at busy hour?
- How much of the existing traffic on the PSTN lines will go over the WAN and become on-net traffic?
- How much actual internet bandwidth will be required?
The only way to avoid costly mistakes is to use actual data from a company’s actual calling patterns. “Rules of Thumb” can be costly! Infortel® Select from ISI Telemanagement Solutions is the ideal tool to help you make these calculations.