Hi Cavan, The time between 183 Session Progress and 200OK may be or may not be controllable. It depends upon the time when user has picked up the phone. If 183 session progress was carrying someother media ( playing tones/announcement etc), then it is the user response which will result in 200OK with SDP and it's time really can not be controlled. User may wait for "x" amount of time after ringing of the phone before he picks up the call and a soft-switch really can not do much to force a user to answer the call within 5-6 seconds.
Could you provide a bit more information about the type of interworking involved here so a more appropriate suggestion can be provide on when the billing should be done. Please do not forget to take inputs from others on this subject before coming to any conclusions :) Regards, Indresh K Singh -----Original Message----- From: Pong Cavan [mailto:[EMAIL PROTECTED] Sent: Thursday, May 19, 2005 6:46 PM To: Singh, Indresh; [email protected] Subject: Re: [Sip-implementors] 183 Session Progress with SDP Hi, Thank you Indresh for your response. I agree with you that we should not be billing early until a connection has been established. During this call, the billing did not start until we (10.1.26.125) sent 200 OK SDP. The thing I would like to understand is why does it take like 23seconds between 183 Session Progress SDP and 200 OK SDP. I would like to shorten this down to 6-10 seconds. Your input is appreciate it! Here's a trace of the call: No. Time Source Destination Protocol Info 1 0.000000 192.168.1.209 10.1.26.125 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Max-Forwards: 30 Session-Expires: 3600;Refresher=uac Supported: timer To: 15552563645 <sip:[EMAIL PROTECTED]> SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.209:5060;branch=c199c936f231b0d8ebbcd61caa81fffe Contact: sip:[EMAIL PROTECTED]:5060 Content-Type: application/sdp Content-Length: 170 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): NexTone-MSW 1234 0 IN IP4 192.168.1.61 Owner Username: NexTone-MSW Session ID: 1234 Session Version: 0 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.1.61 Session Name (s): sip call Connection Information (c): IN IP4 192.168.1.61 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 192.168.1.61 Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 17410 RTP/AVP 18 4 8 0 Media Type: audio Media Port: 17410 Media Proto: RTP/AVP Media Format: ITU-T G.729 Media Format: ITU-T G.723 Media Format: ITU-T G.711 PCMA Media Format: ITU-T G.711 PCMU Media Attribute (a): rtpmap:18 G729/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute Fieldname: fmtp Media Attribute Value: 18 annexb=yes No. Time Source Destination Protocol Info 2 0.000719 10.1.26.125 192.168.1.209 SIP Status: 100 Trying Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.209:5060;branch=c199c936f231b0d8ebbcd61caa81fffe;received=192.168. 1.209;rport=5060 From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 To: 15552563645 <sip:[EMAIL PROTECTED]>;tag=as3ea00078 SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] SIP tag: as3ea00078 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 No. Time Source Destination Protocol Info 3 5.562280 10.1.26.125 192.168.1.209 SIP/SDP Status: 183 Session Progress, with session description Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Status-Code: 183 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.209:5060;branch=c199c936f231b0d8ebbcd61caa81fffe;received=192.168. 1.209;rport=5060 From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 To: 15552563645 <sip:[EMAIL PROTECTED]>;tag=as3ea00078 SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] SIP tag: as3ea00078 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 164 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 29681 29681 IN IP4 10.1.26.125 Owner Username: root Session ID: 29681 Session Version: 29681 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 10.1.26.125 Session Name (s): session Connection Information (c): IN IP4 10.1.26.125 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 10.1.26.125 Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 16214 RTP/AVP 0 Media Type: audio Media Port: 16214 Media Proto: RTP/AVP Media Format: ITU-T G.711 PCMU Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - No. Time Source Destination Protocol Info 4 5.582844 10.1.26.125 192.168.1.61 RTP Payload type=ITU-T G.711 PCMU, SSRC=291861985, Seq=3455, Time=112 Real-Time Transport Protocol No. Time Source Destination Protocol Info 5 5.678392 192.168.1.61 10.1.26.125 RTP Payload type=ITU-T G.711 PCMU, SSRC=141939002, Seq=1245, Time=2651041101 Real-Time Transport Protocol No. Time Source Destination Protocol Info 6 28.942465 10.1.26.125 192.168.1.209 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.209:5060;branch=c199c936f231b0d8ebbcd61caa81fffe;received=192.168. 1.209;rport=5060 From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 To: 15552563645 <sip:[EMAIL PROTECTED]>;tag=as3ea00078 SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] SIP tag: as3ea00078 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 164 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 29681 29682 IN IP4 10.1.26.125 Owner Username: root Session ID: 29681 Session Version: 29682 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 10.1.26.125 Session Name (s): session Connection Information (c): IN IP4 10.1.26.125 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 10.1.26.125 Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 16214 RTP/AVP 0 Media Type: audio Media Port: 16214 Media Proto: RTP/AVP Media Format: ITU-T G.711 PCMU Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - No. Time Source Destination Protocol Info 7 29.013627 192.168.1.209 10.1.26.125 SIP Request: ACK sip:[EMAIL PROTECTED] Session Initiation Protocol Request-Line: ACK sip:[EMAIL PROTECTED] SIP/2.0 Method: ACK Resent Packet: False Message Header Max-Forwards: 30 To: 15552563645 <sip:[EMAIL PROTECTED]>;tag=as3ea00078 SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] SIP tag: as3ea00078 From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.209:5060;branch=c199c936f231b0d8ebbcd61caa81fffe Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 No. Time Source Destination Protocol Info 8 29.498825 192.168.1.61 10.1.26.125 RTP Payload type=ITU-T G.711 PCMU, SSRC=141939002, Seq=1324, Time=2651231661 Real-Time Transport Protocol No. Time Source Destination Protocol Info 9 71.225315 192.168.1.209 10.1.26.125 SIP Request: BYE sip:[EMAIL PROTECTED] Session Initiation Protocol Request-Line: BYE sip:[EMAIL PROTECTED] SIP/2.0 Method: BYE Resent Packet: False Message Header Max-Forwards: 30 To: 15552563645 <sip:[EMAIL PROTECTED]>;tag=as3ea00078 SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] SIP tag: as3ea00078 From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Via: SIP/2.0/UDP 192.168.1.209:5060;branch=0e209cd2eff0e1675752f975b93149f4 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 No. Time Source Destination Protocol Info 10 71.225529 10.1.26.125 192.168.1.209 SIP Status: 200 OK Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.209:5060;branch=0e209cd2eff0e1675752f975b93149f4;received=192.168. 1.209;rport=5060 From: <sip:[EMAIL PROTECTED]:5060>;tag=3325000742-546077 SIP from address: sip:[EMAIL PROTECTED]:5060 SIP tag: 3325000742-546077 To: 15552563645 <sip:[EMAIL PROTECTED]>;tag=as3ea00078 SIP Display info: 15552563645 SIP to address: sip:[EMAIL PROTECTED] SIP tag: as3ea00078 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 ----- Original Message ----- From: "Singh, Indresh" <[EMAIL PROTECTED]> To: "'Pong Cavan'" <[EMAIL PROTECTED]>; <[email protected]> Sent: Thursday, May 19, 2005 4:41 PM Subject: RE: [Sip-implementors] 183 Session Progress with SDP > 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: [email protected] > 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 > [email protected] > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list [email protected] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
