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:[EMAIL PROTECTED] ===========================================
________________________________ From: Jati Kalingga Praja [mailto:[EMAIL PROTECTED] Sent: Friday, January 04, 2008 3:00 AM To: [EMAIL PROTECTED]; [email protected] 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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