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Hi Sushil,
you are touching a problematic area in SIP.
One way with the currently existing standards is to run the call trough a trusted UA of the billing entity. That entity can measure on the RTP (or RTCP) level if the connection is still alive. Typically, this UA will be a PSTN gateway or a B2BUA (back-to-back user agent).
Unfortunately, this model has a lot of drawbacks. It adds additional delay in the media path. This becomes a big problem if there are several billing parties involved in the call, which may lead to absurd delay times (e.g. one billing party located in USA, the other one in India and the next one in Australia). On the other hand, in some situations something like this is needed anyway (symmetrical NAT, legal intercept).
There is currently a proposal to use session timer for keep-alive messages. I like this approach, as the involved user agents might have to answer challenges, making the keep-alive traffic much more authentic. The problem with this approach is the relatively high workload for the SIP network elements (proxies) and that at least one entity has to support session timer (which not always the case today). Also, a guaranteed accounting precision in the second-area can not guaranteed and would rely on a fair usage of the mechanism. However, if you want to reduce your telephone bill by a few seconds, just pull the Ethernet plug when you want to stop talking…
Christian
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