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With traditional Session Initiation Protocol trunking, you plan for your peak bandwidth to support the phones in your voice over IP network. This is different from analog calls, where each user has her own line for direct dialing. Many networks have some combination of analog and SIP; however, SIP tends to be less expensive since it uses an internet connection.
Users with a VPN connection can also use SIP trunks. Some services offer call bursting to handle sudden spikes in concurrent calls. Without the correct configuration, SIP trunks and lack of bandwidth can cause poor call quality in the same way that a weak internet connection can cause poor response times, jitter and delays. Adding users means reevaluating the bandwidth both coming into and leaving the system. If a company provides its own services in and out of the building, then the company is responsible for those connections.
Cloud-based VoIP systems use cloud SIP trunking, and the provider takes responsibility for SIP trunks and bandwidth. Services can be set up in a number of different consumption models, such as a flat rate, per user, per account or some combination of the models.
It doesn't matter where end users are located for cloud SIP trunking, and VPNs aren't necessary as the system can be accessed from virtually anywhere. It also means that system management is likely to be a software dashboard instead of additional hardware. The provider is responsible for determining if all of its users have the right amount of bandwidth needed for their conversations. Offloading the actual systems can save organizations both time and money by shifting management responsibility to their provider, which offers more support resources than having to train and retain resources in-house.
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