RE: [Sip] SIP2Megaco Case:Callee onhook first

["Zhang, Rocky Fan Wen \(Rocky\)" <[e-mail]>]

 
About your second question, you can find help from RFC 3959 and RFC 3960
which tells early media.
Yes, after 180 there can follows RTP packets to let the endpoint to play
remote media.
 
 
 
Best Regards,
 
===========================================
Fanwen zhang
FS5000 CPE IOT, Alcatel-Lucent Qingdao, R&D
Tel: [phone]
Email:[e-mail]
===========================================
 

________________________________

From: Jati Kalingga Praja [mailto:[e-mail]] 
Sent: Friday, January 04, 2008 3:00 AM
To: [e-mail]; [e-mail]
Subject: [Sip] 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. 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 <http://10.14.32.185/>  and 10.14.32.186
<http://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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