[asterisk-biz] Remote SIP monitor

[Jared Smith]

On Wed, 2010-01-06 at 09:45 +0100, Olle E. Johansson wrote:
> I'm adding manager events and storing data in a realtime database -
> one record per call leg. What I'm wondering is how we should handle
> call transfers and hold situations. A call that's transferred has
> multiple streams and RTCP is only valid for one stream.

Would it make more sense to expose the RTCP information as part of the
CEL (Call Event Logging) infrastructure?  That way, you could tell what
events in the call may have triggered the additional streams.  In a
perfect world, we might even have something like:

Event: Incoming call from Alice
Event: Outgoing call to Bob
Event: Asterisk bridges Alice to Bob
Event: RTCP report 
Event: RTCP report
Event: RTCP report
Event: Alice places Bob on hold
Event: RTCP report (new stream, Bob hears hold music from Asterisk)
Event: Unhold
Event: RTPC report (Alice and Bob, again)
Event: Bob transfers Alice to Charlie
Event: RTCP report (new stream, Alice and Charlie)
Event: Hangup

Obviously that's an oversimplified example, but I really think it makes
more sense to put the RTCP reports in the CEL logs, rather than having
another arbitrary log for the data.  Maybe we should move this
discussion to the -dev list to discuss the more technical details?

Anyhoo, just wanted to add my own two cents (US cents, before interest,
taxes, depreciation, and amortization)

--
Jared Smith
Digium, Inc.