List Info

Thread: SIP2Megaco Case:Callee onhook first
SIP2Megaco Case:Callee onhook first
user name
 2008-01-03 12:59:43 

  


  

    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


  

  
  
   
     
    

  

    RE:  SIP2Megaco Case:Callee onhook first
  


  

     country flaguser name
United States		
     2008-01-06 19:50:25
  


  

    

  


  

     

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: 
+86-532-88615493
Email:fanwenzalcatel-lucent.com



===========================================


 






From: Jati Kalingga Praja 
[mailto:king.of.kalinggagmail.com] 
Sent: Friday, January 04, 2008 
3:00 AM
To: megacoietf.org; sipietf.org
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 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 		
  

  
  
   
     
    






 [1-2]











   
 





about | contact  Other archives ( Real Estate discussion Medical topics )