Detailed SIP Call Flow with CVP Comprehensive Model

Introduction
This documents aims to provide detailed SIP CVP comprehensive Call Flow with the debugs captured from the CVP
logs and IOS/VXML Gateways

Network Setup

The setup is very simple to demonstrate the SIP call flow.

 A call comes in from PSTN Phone and goes to the ingress gateway
 Ingress gateway is also acting as VXML Gateway for this setup
 Ingress gateway sends the call to CUP SIP Proxy
 CUP SIP Proxy sends the call to the CVP Call Server
 CVP Call Server will send the call to ICM
 Agents are connected to the CUCM

Following script is being used (Notice that Microapps are being used and no Call Studio script is being used in this
setup)

ICM Script Flow

 Call comes in and hit Start
 Some variables are being set (HTTP Media Server IP Address, Directory Name for the
stored media files, en-us locale, input type being set it digits etc.)
 Then call is being sent to VRU which VXML-GW
 VXML instructions are being sent to the VXML-GW
 Save the digits into a variable
 Play those digits back to the caller
 Set LAA (Longest Available Agent) agent selection algorithm
 Wait in the queue for agent to become available
 During this time play a hold music to the caller in the loop
In the following section the details will be discussed step by step
(1) Call Comes in from the PSTN

PSTNph---PSTN----Ingress-gw----->CUPS
Call matches an incoming pots dial-peer for the destination number. It is important to configure cvp-survivability
service under the POTS dial-peer. If this service is not configured on the incoming pots dial-peer, the ingress gateway
will not be able to communicate with the CVP Call Server and might receive “SIP/2.0 503 Service Unavailable”
message from the CVP Call Server.

dial-peer voice 1 pots

service cvp-survivability
incominca-called-number 5417641400
direct-inward-dial

In the event of some critical CVP application failure, survivability.tcl script will attempt to recover and smoothly
disconnect the caller after playing critical_error.wav file from Ingress-GW’s flash. It is very important to have this
script because this script also plays pleasewait.wav and holdmusic.wav files from flash depending on situation.

Call Matches following outbound sip voip dial-peer on the ingress-gw

dial-peer voice 1400 voip

translation-profile incoming block
preference 1
max-conn 2
destination-pattern 5417641400
session protocol sipv2
session target ipv4:10.4.33.97 ## CUPS SIP PROXY SERVER IP ADDRESS
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad

CUPS load balance the call because there are static routes configured in it and sends call to CVP Call Server

ination VP Server IP Address Priority Weight Protocol

5417641400 10.4.33.131 5060 1 1 UDP
5417641400 10.4.33.132 5060 1 1 UDP
5417641400 10.4.33.133 5060 1 1 UDP

(2) CUPS ----> CVP Server

Call lands to the CVP Call SERVER and in the CVP Call Server log you could see following

[CVP Server Logs]

3052: 10.4.33.131: Mar 13 2008 14:34:39.703 -0700: %CVP_7_0_ICM-7-CALL: {Thrd=Pool:ICM[1450]}
CALLGUID = 8B82AD85F07B11DC80250013192D1650, DLGID = 21 [SIP_LEG] - Publishing ,,
[ICM_NEW_CALL],dialogueId=21, sendSeqNo=1,trunkGroupId=100,trunkNumber=0,serviceId=1,
dialedNumber=5417641400,uui=,, LEGID = 8C1C7CEE-F07B11DC-8177F66C-9B680D18,DNIS =
5417641400,ANI = 4085274003
(3) CVP Server ---> CUICME via VRU-PG

CVP Server sends the call to CUICM via VRU PG. Notice that we didn't configure the IP address of the CUICM or
the VRU-PG in the CVP Call Server. CVP Call Server automatically sends the information to the CUICM's VRU-
PG, because during the CUICM configuration, we had already specified the IP addresses of our CVP Call Servers.
The VRU PG will connect to the ICM Service on the CVP Call Server. Once that TCP socket is established, the CVP
acting as the routing client functionality becomes active in ICM and CVP will communicate with ICM/PG over that
socket.

(4) CUICM [Runs Routing Script]--->Generates Label+Correlation-ID

When call arrives at CUICM, it causes Unified ICM to run a routing script. In the routing script there is a node
called SendToVRU node (Remember: VRU in our case is VXML-GW and SendToVRU node is added right after
setting the basic variables). The SendToVRU is what triggers ICM to transfer the call to the VXML GW (which is
the VRU). The routing script will typically initiate a transfer of the call to a VoiceXML Gateway port (due to
SendToVRU node) via the SIP Service. You could see following in the router process windows on the CUICM
Server

CUICME Router Logs]

14:51:07 Trace: CT_SL_Timer(56)(Init): CT(5000,PreSales_CT), Threshold(20 + 2).
14:51:07 Trace: Correlation id for dialog (23 x 0 : 0 0) is (44).
14:51:07 Trace: Dialog (23 x 44 : 0 0) transfer label = 5417641401

As you can see from above that ICM returns a VRU label 5417641401+correlation-id (44) back to CVP Call Server
(Remember CVP Server was the routing-client).You can see the same information on the CVP Call Server logs that a
call goes to ICM and ICM returns a label back to the CVP Call Server (Which is the routing-client for ICM). This
correlation-id (which is 44 in our case ) will be appended at the end of the label.

[CVP Call Server Logs]

3053: 10.4.33.131: Mar 13 2008 14:34:39.718 -0700: %CVP_7_0_ICM-7-CALL: {Thrd=Pool:ICM[1451]}
CALLGUID = 8B82AD85F07B11DC80250013192D1650, DLGID = 21 [SIP_LEG] - Processing ,,
[ICM_TEMPORARY_CONNECT], dialogueId=21,sendSeqNo=1, label=5417641401, correlationId=44,,
LEGID = 8C1C7CEE-F07B11DC-8177F66C-9B680D18, DNIS = 5417641400, ANI = 4085274003

Now CVP Call Server will send this VRU label (5417641401+44) to the VXML GW (It is more accurate to say that
CVP transfers the call TO the VXML GW, with the label+correlation ID as the destination phone number). In our
case, it is only coincidence that the VXML GW is the same as the ingress GW. The “transfer” that CVP does is not a
typical one either, because CVP stays in the call path (whereas usually the transferor drops out of the call path).
In reality CVP is setting up a NEW call to the VXML GW and is shuffling the SDP from the ingress GW to the
VXML GW in order to connect the media directly (similar to CUCM or CUBE in flow-around mode). CVP stays in
the signaling path so that it can retain call control.
[VXML-GW Debugs]
ebug voip application vxml
calling Number=sip:4085274003@10.4.33.131:5060,(Calling Name=)(TON=National, NPI=ISDN,
Screening=User, Passed, Presentation=Allowed),
Called Number=sip:541764140144@10.4.33.1:5060;transport=tcp(TON=Unknown, NPI=Unknown)

This label matches 1401 voip dial-peer in the VXML-GW and invokes bootstrap VXML application.

dial-peer voice 1401 voip

translation-profile incoming block
service bootstrap
 session protocol sipv2
incoming called-number 5417641401T ## T is used to match the correlation-id
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad

The bootstrap VXML application tells the CVP that it is a new call and CVP then sends the same label
(5417641401+44) back to ICM. Once ICM sees its own generated label, this actually completes the loop and ICM at
this point knows that there is a VXML Gateway ready to accept the instructions.

(5) CUICM---> RunExternalScript node

The steps below are the RunExternalScript nodes that tell CVP what VXML pages to send to the VXML gateway
(play prompt, get digits, etc). CUICM execute the script instruction called “RunExternalScript”. This basically tells
CVP that IVR related instructions are coming. CUICM then sends instructions like for example “Get4Digits”. CVP
sends this instruction to VXML-GW to play the prompt to the caller. When caller enters the digit, ICM stores it in its
variable [for example Call.CallerEnteredDigits variable].

[VXML-GW Debugs]
063686: Mar 13 18:25:14.383 PST: //404/vapp_digit_collection_done: digits [5432], status [0], pattern [v0]
063687: Mar 13 18:25:14.383 PST: //404/vxml_digit_collection_done:
vxmlp 6A6D1EA4 status 0 async_status 200000000
063688: Mar 13 18:25:14.383 PST: //404/vxml_digit_collection_done: digits (5432)

ICM Script then instructs CVP to play the prompts back to the user on the PSTN side. For simplicity we can say that
in order to do that CVP makes an inbound call to the VXML gateway by dialing 987654 and plays the prompt. But
in reality we wouldn’t say that CVP makes an inbound call to the VXML gateway (since that already happened
above). The 987654 is actually a transfer instruction that was received via VXML from CVP, the only reason it exists
is because it is a workaround for some problem with playing media files and retaining the vxml session.
[VXML-GW DEBUGS]

063767: Mar 13 18:25:14.407 PST: //404/vxml_item_attrs_proc:

name=mycall
URI(abs):phone://987654
scheme=phone
host=987654 bridge=1 connecttimeout=0 maxtime=0 desttype=-1 destplan=-1
anitype=-1 aniplan=-1 anipi=-1, anisi=-1 rdntype=-1 rdnplan=-1 rdnpi=-1, rdnsi=-1, redirectreason=-1

063779: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/5.wav:

063780: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/4.wav:
063781: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/3.wav:
063782: Mar 13 18:25:14.407 PST: //404/vapp_media_play: prompt=http://10.4.33.130/en-us/sys/2.wav:

(6) CUICM---->Agent

 When an agent becomes available, ICM instructs CVP to transfer the call to the agent. CVP will disconnect
the call to the VXML GW (SIP BYE), will send a re-INVITE back to the ingress gateway to stop media,
and will send a new INVITE to the agent phone and to the ringtone service below.

 When the ringtone service answers, a re-INVITE is sent to the ingress gateway with the
 VXML GW’s SDP in order to establish the media stream for the ringback.

 Once the Agent answers the phone, CVP disconnects the call to the VXML GW (with BYE & re-INVITE
same as before), and then connects the media stream from the ingress gateway to the agent phone.

 In this entire call flow, there have been 4 distinct SIP phone calls that are separate from each other:

o ingress gateway to CVP;

o CVP to VXML GW (IVR);
o CVP to VXML GW (ringtone);
o CVP to Agent.

 Each phone call, from a SIP perspective, is completely separate from the others, because CVP is a B2BUA
and not a proxy.

 Whenever CVP does a transfer, it disconnects the previous destination that the caller was connected to and
connects them to the new destination.

 Thus, CVP acts as a pivot point for the call.

 Since CVP is under control of ICM, that is what makes CVP so powerful – the ability to control the call
from very first component that received the call. This is in contrast to a typical PBX/ACD (including CUCM)
where the transfers are tromboned through every component that was involved in the call, tying up additional
DS0’s or memory.

 In a SIP call flow, during the time when CUICM is transferring call over to the agent, there is a need to play
ringtone back to the caller. And also if in case there is an error during transfer, CVP should be able to play error
tone back to the caller. These tones will be generated by playing a .wav file off of the VXML gateway flash.

 In the CVP, there are two parameters under SIP section. These are the DNs that CVP sends to VXML GW
when there is a need to generate the ringtone or error tone.

 When these numbers reach the VXML GW, it matches with an incoming voip dial-peer and VXML GW
starts the respective service/app
dial-peer voice 9191 voip
service ringtone
incoming called-number 91919191
dtmf-relay rtp-nte h245-signal h245-alphanumeric
codec g711ulaw
session protocol sipv2
no vad
!
dial-peer voice 9292 voip
service cvperror
incoming called-number 92929292
dtmf-relay rtp-nte h245-signal h245-alphanumeric
codec g711ulaw
session protocol sipv2
 no vad

 You can see the VXML-GW playing the file by noticing the following debugs

[Debugs VXM GW]

062151: Mar 13 18:14:12.483 PST: //383/tcl_MediaObjCmd: media play leg_incoming flash:ringback.wav
062152: Mar 13 18:14:12.483 PST: //383/tcl_MediaPlayObjCmd: play leg_incoming flash:ringback.wav

NOTE: Ignore the timestamp in the debug messages. These debugs were captured by making multiple calls over a
period of time
Different Gateway Applications

bootstrap.tcl
The script extracts required parameters and then hands off the call to the boot strap VXML service named "new-call"
which loads the bootstrap.vxml file. This VXML page then submits a new call HTTP request to the Customer Voice
Portal (CVP) IVR Service which kicks off a sequence of VXML communications between the VXML Gateway and
the CVP IVR Service, commonly called "MicroApps". For SIP calls, this script extracts Call GUID, Call Server
URL, and DNIS. Call Server host is extracted from the App-Info header. For H.323 calls, Call GUID and DNIS are
extracted. Call Server is also extracted from the call data. For both SIP and H.323 calls, the Call GUID is normalized
in base16, 35 char length hex format as follows(not base10, 43 char length):
HHHHHHHH-HHHHHHHH-HHHHHHHH-HHHHHHHH

bootstrap.vxml
The VXML-GW sends a new call request to the Customer Voice Portal (CVP) IVR Service which kicks off a
sequence of VXML communications between the VXML Gateway and the CVP IVR Service, commonly called
“MicroApps”. Any HTTP errors, or fetch errors (7 second timeout), will handoff to recovery.vxml in flash and
handoff.tcl in flash.

Handoff.tcl
Handoff.tcl has the sole job of disconnecting a call with a code an ISDN Q.850 code of 38.

survivablity.tcl
This service should be placed on the ingress-gw on the incoming pots dial-peer that is destined for CVP. In the event
of critical CVP application errors or a WAN failure that would normally disconnect the caller, this script allows the
gateway to attempt a transfer to some alternate location after the failure occurs instead of disconnecting the caller. In
the event that the call cannot be transfered to an alternate agent, the script will play a "call-back-later" message and
disconnect.