SIP-Based Admission Control | Voice Mobility with Wi-Fi

The first method is to rely on the call setup signaling. Because the most common mechanism today is SIP, we can refer to this as SIP-based admission control. The idea is fairly simple. The access point, most likely in concert with a controller if the architecture in use has one, uses a firewall-based flow-detection system to observe the SIP messages as they are sent from the phones to the SIP servers and back. Specifically, when the call is initiated, either by the phone sending a SIP Invite, or receiving one from another party, the wireless network determines whether there is available capacity to take the call. If there is available capacity, then the wireless network lets the messages flow as usual, and the call is initiated.
On the other hand, if the wireless network determines that there is no room for the call, it will intercept the SIP Invite messages, preventing them from reaching the other party, and interject its own message to the caller (as if from the called party, usually), with one of a few possible SIP busy statuses. The call never completes, and the caller will get some sort of failure message, or a busy tone.
Other, more advanced behaviors are also possible, such as performing load balancing, once the network has determined that the call is not going to complete.
The advantage of using SIP flow detection to do the admission control is that it does not require any added sophistication on the mobile devices than they would already have with SIP. Furthermore, by having that awareness from tracking the SIP state, the network can provide a list of both calls in progress and registered phones not yet in a call. The disadvantage is that this system will not work for SIP calls that are encrypted end-to-end, such as being carried over a VPN link.

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