SIP2Megaco Case:Callee onhook first
"Jati Kalingga Praja" <[email protected]>
| Newsgroups | gmane.ietf.megaco,gmane.ietf.sip |
|---|---|
| Message-ID | <[email protected]> |
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 _______________________________________________ Megaco mailing list [email protected] https://www1.ietf.org/mailman/listinfo/megaco