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


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