It depends upon what is carried in the 183 SDP. Let us say 183 Is carrying a SDP which connects A to a Media Server and Media Server is just playing an announcement, that your call is proceeding. In that case you would not want to start billing that person after receiving media in 183.
200 OK SDP generally carries the end user's SDP providing the confirmation that the user has accepted the call and is initiating the conversation, so that is the point of time when the billing should start. This is applicable for the case of interworking too, but sometimes at the time of sending 183 the SDP indicates that user has accepted the call, so I think if you provide more detail regarding what SDP is being carried in 183 what is actually happening at the remote end ( Say it is PRI/ISUP/H323/MGCP then what is the level of signaling on the other side, whether at the point of sending 183 User has picked up the phone or not ). one may provide more appropriate suggestion. Billing generally starts when speech path is cut through and speech path to the end-user is cut through normally after 3-way handshake of INVITE 200OK ACK Txn is completed. In between if say 183 carries SDP, then it will depend upon what SDP it carries and whether speech path is being cut through to the end user or to something else. If it is being cut through to the end user, it makes sense to start billing immediately otherwise not. Regards, Indresh K Singh -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Pong Cavan Sent: Thursday, May 19, 2005 4:13 PM To: sip-implementors@cs.columbia.edu Subject: [Sip-implementors] 183 Session Progress with SDP Dear Sirs, I am a newbie and please forgive me if this post does not below in this list. I have a question that I hope you might be able to clarify for me. Gateway A sends an INVITE to Gateway B with SDP. When B sends back 183 Session Progress with SDP, shouldn't A respond and use the information within the 183 SDP instead of waiting for B's 200 OK SDP? The cdr shows the duration of the call as 72 seconds and the billable second as 43. That is almost 29 seconds before the call is picked up. Shouldn't the 183 SDP from B to A help shorten this post dial delay? Thank you very much for your time! Regards, Pong 192.168.1.209 (A) 10.1.26.125 (B) 192.168.1.61 (A's Media Gateway) | | | | | | 0.000 |INVITE SDP (g729 g711U)| | |------------------------------------>| | | | | 0.001 | 100 Trying | | |<------------------------------------| | | | | 5.562 |183 Session Progress SDP (g711U) | |<------------------------------------| | | | | | | RTP (g711U) | 5.583 | |-------------------->| | | | | | RTP (g711U) | 5.678 | |<--------------------| | | | 28.942 | 200 OK SDP (g711U) | | |<------------------------------------| | | | | 29.014 | ACK | | |------------------------------------>| | | | | 29.499 | | RTP (g711U) | | |-------------------->| 71.225 | BYE | | |------------------------------------>| | | | | 71.226 | 200 OK | | |<------------------------------------| | _______________________________________________ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
