- Cisco Unified Border Element Protocol-Independent Features and Setup
- Interworking Between RSVP Capable and RSVP Incapable Networks
- SIP INFO Method for DTMF Tone Generation
- WebEx Telepresence Media Support Over Single SIP Session
- DTMF Events through SIP Signaling
- Call Progress Analysis Over IP-to-IP Media Session
- Codec Preference Lists
- AAC-LD MP4A-LATM Codec Support on Cisco UBE
- Multicast Music-on-Hold Support on Cisco UBE
- Network-Based Recording
- Video Recording - Additional Configurations
- TDoS Attack Mitigation
- Cisco Unified Communications Gateway Services--Extended Media Forking
- Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls
- Acoustic Shock Protection
- Noise Reduction
- iLBC Support for SIP and H.323
- Configuring RTP Media Loopback for SIP Calls
- SIP Ability to Send a SIP Registration Message on a Border Element
- Session Refresh with Reinvites
- SIP Stack Portability
- Interworking of Secure RTP calls for SIP and H.323
- Cisco UBE Support for SRTP-RTP Internetworking
- Support for SRTP Termination
- Configuring RTCP Report Generation
- SIP SRTP Fallback to Nonsecure RTP
- Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks
- VoIP for IPv6
- Mid-call Signaling Consumption
- Support for Software Media Termination Point
- Cisco Unified Communication Trusted Firewall Control
- Cisco Unified Communication Trusted Firewall Control-Version II
- Domain-Based Routing Support on the Cisco UBE
- URI-Based Dialing Enhancements
- Fax Detection for SIP Call and Transfer
- Feature Information for Call Progress Analysis Over IP-IP Media Session
- Restrictions for Call Progress Analysis Over IP-to-IP Media Session
- Information About Call Progress Analysis Over IP-IP Media Session
- How to Configure Call Progress Analysis Over IP-to-IP Media Session
- Configuration Examples for the Call Progress Analysis Over IP-to-IP Media Session
Call Progress Analysis Over IP-to-IP Media Session
The Call Progress Analysis Over IP-IP Media Session feature enables the detection of automated answering systems and live human voices on outbound calls and communicates the detected information to the external application. Typically, call progress analysis (CPA) is extensively used in contact center deployments in conjunction with the outbound Session Initiation Protocol (SIP) dialer, where CPA is enabled on the Cisco Unified Border Element (Cisco UBE), and digital signal processors (DSP) perform the CPA functionality.
- Feature Information for Call Progress Analysis Over IP-IP Media Session
- Restrictions for Call Progress Analysis Over IP-to-IP Media Session
- Information About Call Progress Analysis Over IP-IP Media Session
- How to Configure Call Progress Analysis Over IP-to-IP Media Session
- Configuration Examples for the Call Progress Analysis Over IP-to-IP Media Session
Feature Information for Call Progress Analysis Over IP-IP Media Session
The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to . An account on Cisco.com is not required.
Feature Name |
Releases |
Feature Information |
---|---|---|
Call Progress Analysis Over IP-to-IP Media Session |
15.3(2)T |
The Call Progress Analysis Over IP-to-IP Media Session feature enables detection of automated answering systems and live human voices on outbound calls and communicates the detected information to an external application. The following command was introduced: call-progress-analysis. |
Call Progress Analysis Over IP-to-IP Media Session |
Cisco IOS XE Release 3.9S |
The Call Progress Analysis Over IP-to-IP Media Session feature enables detection of automated answering systems and live human voices on outbound calls and communicates the detected information to an external application. The following command was introduced: call-progress-analysis. |
Support for additional call flows |
15.5(2)T Cisco IOS XE Release 3.15S |
|
Restrictions for Call Progress Analysis Over IP-to-IP Media Session
-
Only SIP-to-SIP Early Offer (EO-to-EO) call flows are supported.
-
Session Description Protocol (SDP) passthrough and flow-around media calls are not supported.
-
Only the G711 flavor of codec is supported.
-
High Availability (HA) is not supported.
-
Skinny Client Control Protocol (SCCP)-based digital signal processor (DSP) farm is not supported.
-
CPA cannot not be detected if Dialer uses Inband as DTMF relay mechanism, that is, Inband to RTP-NTE DTMF inter-working is not supported with CPA.
-
CPA call record is not supported for "180 without SDP" and "Direct Call Connect (without 18x)" call flows from Service Provider.
Information About Call Progress Analysis Over IP-IP Media Session
Call Progress Analysis
Call progress analysis (CPA) is a DSP algorithm that analyzes the Real-Time Transport Protocol (RTP) voice stream to look for special information tones (SIT), fax or modem tones, human speech, and answering machine tones. CPA also passes the voice information to Cisco IOS or Cisco Unified Border Element (Cisco UBE).
CPA is initiated on receiving a new SIP INVITE with x-cisco-cpa content. While a call is in progress, the DSP or the Xcoder analyzes the incoming voice or media stream. The DSP identifies the type of voice stream based on statistical voice patterns or specific tone frequencies and provides the information to the Cisco UBE. The Cisco UBE notifies the dialer with a SIP UPDATE with x-cisco-cpa content along with the detected event. Based on the report, the caller (dialer) can decide to either transfer the call or terminate the call.
To use the CPA functionality, you must enable CPA and configure CPA timing and threshold parameters.
SIP Message |
Direction of Message |
Meaning |
---|---|---|
18x or 200 |
Cisco IOS to dialer |
Cisco UBE informs the dialer if CPA is enabled for a call or not. |
New INVITE |
Dialer to Cisco IOS |
Dialer requests Cisco IOS or the Cisco UBE to activate the CPA algorithm for this session. |
UPDATE |
Cisco IOS to dialer |
Cisco IOS or the Cisco UBE notifies the dialer about the detected event. |
CPA Events
CPA Event |
Definition |
---|---|
Asm |
Answer machine |
AsmT |
Answer machine terminate tone |
CpaS |
Start of the Call Progress Analysis |
FT |
Fax/Modem tone |
LS |
Live human speech |
LV |
Low volume or dead air call |
SitIC |
Special information tone IC -- Intercept -- Vacant number or Automatic Identification System (AIS) |
SitNC |
SIT tone NC—No Circuit (NC), Emergency, or Trunk Blockage |
SitVC |
SIT tone VC—Vacant Code |
SitRO |
SIT tone RO—Reorder Announcement |
SitMT |
Miscellaneous SIT Tone |
How to Configure Call Progress Analysis Over IP-to-IP Media Session
- Enabling CPA and Setting the CPA Parameters
- Verifying the Call Progress Analysis Over IP-to-IP Media Session
- Troubleshooting Tips
Enabling CPA and Setting the CPA Parameters
Perform the following task to enable CPA and set the CPA timing and threshold parameters:
1.
enable
2.
configure terminal
3.
dspfarm profile profile-identifier transcode
4.
call-progress-analysis
5.
exit
6.
voice service voip
7.
cpa timing live-person max-duration
8.
cpa timing term-tone max-duration
9.
cpa threshold active-signal signal-threshold
10.
end
DETAILED STEPS
Verifying the Call Progress Analysis Over IP-to-IP Media Session
Perform this task to verify that call progress analysis has been configured for a digital signal processor (DSP) farm profile.
1.
enable
2.
show dspfarm profile profile-identifier
DETAILED STEPS
Troubleshooting Tips
Use the following commands to troubleshoot the call progress analysis for SIP-to-SIP calls:
Configuration Examples for the Call Progress Analysis Over IP-to-IP Media Session
Example: Enabling CPA and Setting the CPA Parameters
The following example shows how to enable CPA and set a few timing and threshold parameters. Depending on your requirements, you can configure more timing and threshold parameters.
Device> enable Device# configure terminal Device(config)# dspfarm profile 15 transcode Device(config-dspfarm-profile)# call-progress-analysis Device(config-dspfarm-profile)# exit Device(config)# voice service voip Device(conf-voi-serv)# cpa timing live-person 2501 Device(conf-voi-serv)# cpa timing term-tone 15500 Device(conf-voi-serv)# cpa threshold active-signal 18db Device(conf-voi-serv)# end