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[Sip] SIP2Megaco Case:Callee onhook first

"Jati Kalingga Praja" <king.of.kalingga@gmail.com> Thu, 03 January 2008 18:59 UTC

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Date: Fri, 04 Jan 2008 01:59:43 +0700
From: Jati Kalingga Praja <king.of.kalingga@gmail.com>
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Subject: [Sip] SIP2Megaco Case:Callee onhook first
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Hi,,

Now, i have questions too in a call from SIP phone to analog phone which the
case was the callee onhooked first, which is the analog phone. You can
download captured data in this url:
http://www.4shared.com/file/33781592/42539b99/4577020ke4580220ttupdtlpn.html.
And you can view the data using this filter : *(megaco and
ip.addr==10.14.32.186) or (sip and ip.addr==10.14.32.185) or rtp.

*The questions are:
1. When the callee onhooked, why did the MGC send INVITE to the SIP Phone
(line 3845)?
2. Is that true, that the ring back tone, busy tone, dial tone can be sent
using RTP? I mean, why did after 180 Ringing (line 1621), RTP was sent
between 10.14.32.185 and 10.14.32.186? Did the RTP contain ring tone?If yes,
is there a standard explain this?

That's all that i want to ask,,,I hope you guys can help me.
Thx in advance before.

Regards,

Jati

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