 Introduction

 Network Setup
 ICM Script Flow
 (1) Call Comes in from the PSTN
 (2) CUPS ----> CVP Server
 (3) CVP Server ---> CUICME via VRU-PG
 (4) CUICM [Runs Routing Script]--->Generates Label+Correlation-ID
 (6) CUICM---->Agent
 Different Gateway Applications

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
 

Destination         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]
 

debug voip application vxml

calling Number=sip:[email protected]:5060,(Calling Name=)
(TON=National, NPI=ISDN,
Screening=User, Passed, Presentation=Allowed),
Called Number=sip:[email protected]: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