[Sip] RE: [Megaco] SIP2Megaco Case:Callee onhook first

"John Wainwright" <john.wainwright@txpcorporation.com> Fri, 04 January 2008 14:42 UTC

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From: "John Wainwright" <john.wainwright@txpcorporation.com>
To: "Jati Kalingga Praja" <king.of.kalingga@gmail.com>, <megaco@ietf.org>, <sip@ietf.org>
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Subject: [Sip] RE: [Megaco] SIP2Megaco Case:Callee onhook first
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Tones can be played over the RTP stream.  Typically a tone is defined as
proceeding from the Termination towards the exterior of the context so
if a tone is requested to be played on a Phy termination it is
played/generated locally while if it is played to the EPH termination it
is played on the RTP stream.

 

See megaco standards document section 7.1.11 ( V2 document )

 

John

 

 

________________________________

From: Jati Kalingga Praja [mailto:king.of.kalingga@gmail.com] 
Sent: Thursday, January 03, 2008 12:25 PM
To: megaco@ietf.org; sip@ietf.org
Subject: [Megaco] SIP2Megaco Case:Callee onhook first

 

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. Here,
i attached captured data too. 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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