Pong,
I was going to ask you why the long delay between the 183 and 200 !,
I guess you beat me to it.
The trace is helpful but doesn't explain the delay. The field "Resent
Packet: False" might not mean much depending on where the trace was taken
and over what transport protocol the signalling was sent. But the network
would have to have lost a lot of 200s to account for any significant amount
of the 23seconds.
It is more likely that this is processing delay in the Asterisk PBX
and / or PSTN signalling components. For the complete picture here you need
to know what PSTN signalling events occurred for the same call. The 200 OK
is not generated and sent until the PSTN ANM (answer)message - which may
have taken the full 28+seconds to be recived.
The delay can be further explained by what PSTN CPG (call progress)
messages were recieved and also what if anything was sent on the early
audio from the PSTN. Did the end user here any announcements in this
scenario or ringing up until the call completed (receiving end to end
audio)?.
illustration (not saying this is what happened below!) - network might have
terminated on an IVR at the 5sec mark, sent back an Address Complete (ACM)
message Atserisk mapped to a 183 with SDP for early media which should be
cut-through so the annoucement is heard by the end party. The message plays
for 15seconds saying if you didn't press 1 or 2 it would put you through to
the operator, you don't so it does, and at 28second mark PSTN sends a ANM
back and call completes end to end.
Hope the above is at least a little helpful - Wayne.
Pong asked:
***********************************
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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