Recording Conversations a network solution

Size: px
Start display at page:

Download "Recording Conversations a network solution"

Transcription

1

2 Recording Conversations a network solution George Gary, Product Manager Updated: October 2013

3 Abstract Intelligent network-based Recording, a new feature in Cisco Unified Communications Manager (CUCM) release 10. Learn how Unified Communications Manager release 10 intelligently instructs voice gateways, Unified Border Elements, and IP Phones to route conversation media to recording servers like MediaSense located anywhere in the network. Capture mobile calls and centralize multi-cluster recording. Participants will learn how to configure Recording with recommended best practices to help ensure a successful deployment. Geared towards partners, customers, and telephony architects. A working knowledge of Unified CM is recommended. 3

4 Agenda Feature Overview Call flows & Use Cases Meta-Data Serviceability & Redundancy Performance Setting it up Usage and Integration Best Practices Partners Recording secure calls 4

5 Recording Conversations

6 Recording History Silent Recording - introduced in Unified Communications Manager release 6.0 Cisco Phones are intelligently instructed to send copies of conversations to Recorders Secure Recording in release 8.0 Meta-data enhanced in release 8.6(2) User Controls for Recording in release 9.0 Network Recording in release 10.0 Combinations of Gateways and Phones are intelligently instructed to send copies of conversations to recorders based on: Call Flows, Participants, and Media requirements 6

7 Network-based Recording Features & Benefits Capture and store conversations for review, analysis, and legal compliance Record calls regardless of device, location, or geography Calls extended off-network to Mobile and Home Office phones Centralize recording policy Automatically selects the right media source based on call flow and call participants. Meta-Data enables applications to track recorded calls in Single-and Multi-cluster environments Delivers 2 unmodified RTP streams to the recording server via SIP Trunk 7

8 Network-based Recording Features & Benefits Provides Call Admission Control, Bandwidth Reservation, and Codec Negotiation Play notification tones when legal compliance is required Serviceability counters and alarms help Compliance officers ensure calls are recorded by monitoring the real-time status and historical performance of the feature Network topology friendly Does not require SPAN Any conversation may be recorded; does not require contact center software 8

9 Network Recording How does it work?

10 Recording Definitions Calling party person initiating the call; also referred to as Customer Called party person answering the call; also referred to as Agent Recorder application responsible for capturing and storing conversation media Recording Media Source the entity (Phone or Gateway) selected to copy and forward conversation media to the Recorder Built-in-Bridge (BiB) the Cisco IP Phone resource which copies and forwards the media streams of the Calling and Called parties (conversation media) to the Recorder Gateway the Cisco Voice Gateway or Unified Border Element which copies and forwards conversation media to the Recorder 10

11 Recording Media Source Administrator specifies their preference to use either the Phone or Gateway to copy and forward conversation media to the recorder Default is Phone Preferred If the call flow, participants, or media requirements change during the call, Unified CM automatically changes the recording media source as needed 11

12 Gateway Preferred Recording How does it work? does it work? A call is received and answered on a line configured for automatic silent recording Cisco Unified CM automatically sends two call setup messages to the gateway 1 st call is for Called Party stream 2 answer Unified CM 3 setup 4 setup PSTN 1 dial 2 nd call is for Calling Party stream Unified CM INVITES Recorder to both calls via SIP Trunk Recorder accepts both calls and receives two RTP streams from the gateway Called Recorder 5 invite 6 invite Calling An optional recording tone can be configured to play to the Calling, Called, or both parties Recording tone settings overrides monitoring tone settings when the call is being monitored and recorded simultaneously Signaling Called stream Calling stream 12

13 Phone Preferred Recording How does it work? A call is received and answered on a line configured for automatic silent recording Unified CM 1 dial Cisco Unified CM automatically sends two call setup messages to the BiB (Called) device 1st call is for Called Party stream 2nd call is for Calling Party stream 2 answer PSTN Calling Unified CM INVITES Recorder to both calls via SIP Trunk Recorder accepts both calls and receives two RTP streams from the phone BiB An optional recording tone can be configured to play to the Calling, Called, or both parties Recording tone settings overrides monitoring tone settings when the call is being monitored and recorded simultaneously Called 3 setup 4 setup 5 invite 6 invite Recorder Signaling Called stream Calling stream 13

14 Recording Architecture Gateway selected TAPI/JTAPI Spervisor s Desktop agent1 call1 [monitor] agent3 idle [monitor] Gateway selected as recording media source : Gateway relays two independent media streams (Calling + Called) to the Recorder Calling (Customer) Caller s voice stream Agent s voice stream PSTN/WAN voice gateway SIP SIP/SCCP Unified CM TAPI/JTAPI Called Recorded User SIP Trunk Recording enabled desktop application Recorder 14

15 Recording Architecture Phone selected TAPI/JTAPI Spervisor s Desktop agent1 call1 [monitor] agent3 idle [monitor] Phone selected as recording media source: Called phone relays two independent media streams (Calling + Called) to the Recorder Calling (Customer) Caller s voice stream Agent s voice stream PSTN/WAN voice gateway MGCP/H323 SIP/SCCP Unified CM TAPI/JTAPI Called Recorded User SIP Trunk Recording enabled desktop application Recorder 15

16 How is the recording source selected? Administrator preference + media type + call flow Preferred Recording Source Media Type Gateways in Call Flow? Selected Recording Source Gateway Phone Unsecure RTP Secure srtp Unsecure RTP Secure srtp Yes Gateway (2) No Phone (3) Yes Phone (4) No Phone (4) Yes Phone (1) No Phone (1) Yes Phone (1) No Phone (1) 1. Phone in call flow, media type is supported, phone is preferred; phone is selected. 2. Gateway in call flow, media type is supported, gateway is preferred; gateway is selected 3. Gateway not in call flow; phone is selected 4. Gateway does not support secure media (srtp) recording; phone is selected 16

17 Gateway preference honored Unsecure media, Gateway in call flow Unified CM Signaling Called stream Calling stream PSTN Called 2 RTP 3 4 RTP 1 Calling Recorder [1] [3]: An external call is answered by a User with a Cisco IP phone [4]: The recording session is automatically started from IP phone; the recording media source is Gateway Preferred and is valid for the call flow; Gateway is selected to fork the media 17

18 Phone preference honored Unsecure media, Phone in call flow Unified CM Signaling Called stream Calling stream PSTN Called 2 RTP 4 3 RTP 1 Calling Recorder [1] [3]: An external call is answered by a User with a Cisco IP phone [4]: The recording session is automatically started from IP phone; the recording media source is Phone Preferred and is valid for the call flow; IP Phone is selected to fork the media 18

19 Gateway preference NOT honored Unsecure media, Gateway not in call flow Calling 1 RTP 3 Unified CM PSTN Signaling Called stream Calling stream Called 2 RTP 4 Recorder [1] [3]: An internal call is answered by a User with a Cisco IP phone [4]: The recording session is automatically started from the Called user IP Phone; the recording media source can be Phone Preferred or Gateway Preferred, but gateway is not valid for the call flow; IP Phone is selected to fork the media 19

20 Alternate Source Selection If the selected source is not available (or valid), Unified CM automatically attempts to use an alternate source to help ensure the call is recorded When gateway is selected, Unified CM attempts to use the first gateway in the call flow, which may be the ingress or egress gateway. If the first gateway is not available, the last gateway is selected. If neither gateways are available, the phone is selected. When phone is selected, Unified CM attempts to use the phone. If the phone source is not available, UCM will attempt to use the first gateway in the call flow, which may be the ingress or egress gateway. If the first gateway is not available, the last gateway is attempted. Selection Order Gateway Selected Phone Selected 1 First Gateway in call path Phone 2 Last Gateway in call path First Gateway in call path 3 Phone Last Gateway in call path 20

21 Multiple gateways in call flow Unified Mobility example - Single Number Reach An external call is received for Alice. Alice is not in the office, so the call is extended off-network where it is answered on her mobile phone. Gateway 1 is the first gateway in the call path and supports recording, so Gateway 1 is selected Recorder Unified CM PSTN PSTN Caller Gateway 1 Alice s Desk phone Gateway 2 Alice answers call on her mobile phone 21

22 Multiple gateways in call flow Unified Mobility example - Single Number Reach An external call is received for Alice. Alice is not in the office, so the call is extended off-network where it is answered on her mobile phone. Gateway 1 is the first gateway in the call path, but does not support recording, so Gateway 2 is selected Recorder Not enabled for recording Unified CM PSTN PSTN Caller Gateway 1 Alice s Desk phone Gateway 2 Alice answers call on her mobile phone 22

23 Automatic & Selective Recording Automatic recording - Records all calls on a line appearance. Invoked by Unified CM. No visual indication of recording session. Selective recording - Allows calls to be recorded as needed Selective Silent Recording - Invoked by Supervisor via CTI-enabled desktop or Recording server based on business rules and events. No visual indication of recording session. Selective User Recording - Invoked by Agent via CTI-enabled desktop and/or Softkey/Programmable Line Key. Cisco IP phone displays recording session messages. Available in Unified CM 9.0(1) or later 23

24 Selective Silent Recording Also works when Gateway is recording media source Supervisors can start/stop recording sessions from CTIenabled desktops Training Quality assurance Configure recording business rules to capture conversations based on policy International Emergency Supervisor Desktop agent1 on call1 [monitor] agent2 idle [monitor] agent3 idle [monitor] Calling Unified CM 3 setup 4 setup Called Called Calling JTAPI/TAPI Recorder 5 invite 6 invite Supervisor OR 24

25 Selective User Recording Selective User Recording is also supported when Gateway is recording media source 1 dial 2 answer Users can start/stop recording sessions from their Cisco IP Phone Calling Called 3 setup 4 setup 5 invite 6 invite Calling Called Recorder Called Calling

26 Silent & User Selective Recording Modes Interaction Silent recording is the default selective recording mode. No visual recording session messages are displayed on Cisco IP phone User recording is available in Unified CM release 9.0(1) or later Provides visual message on Cisco IP device display indicating when a recording session is inprogress User can start/stop recording session from Cisco IP device via softkey/programmable line key and/or CTI-enabled application Mobility Users can start/stop recording sessions via DTMF (*86). Available in release 10.0 or later Silent & User Selective recording modes may not be used together When a silent recording session in in progress, a user recording session cannot be started. User may see display message Recording already started when attempting to start a User recording session. When a user recording session in in progress, a silent recording session cannot be started. Supervisor may see Recording already started when attempting to start a Silent recording session. Automatic recording is always silent 26

27 Codec Selection When Phone is selected as the recording media source, the recording codec is always the same as the original codec for the duration of the call When Gateway is selected as the recording media source, the first recording codec used is the same as the original call If the original conversation codec changes, the codec is dynamically re-negotiated If region settings for the recorder require different codecs, a transcoder is automatically inserted 27

28 Recording Notification Tones For legal compliance, an explicit notification in the form of a periodic tone can be made audible to Called, Calling, or both parties to indicate a recording session is in progress. The tone can also be disabled. Recording tone settings override monitoring tone settings when both are enabled for the same call Notification Tone configured to play to: Called Hears Calling Hears Recording stream for Called Recording stream for Calling No One Nothing Nothing Nothing Nothing Called Party Tone Nothing Nothing Nothing Calling Party Nothing Tone Tone Nothing Both Tone Tone Tone Tone 28

29 Recording Call Flows Single Cluster IP Phone Including Mid-Call features: Hold/Resume, Shared-lines Transfer Conference Automatic recording mode is used for illustrations. Selective recording modes are supported unless otherwise noted.

30 Phone selected as recording source Recording Enabled Gateway Phone Preferred Recording TDMgateway/ CUBE IP phone Recorder [1] [3]: An external call is answered by a User with a Cisco IP phone [4]: The recording session is automatically started from IP phone; the recording media source is Phone Preferred and is valid for the call flow; IP Phone is selected to fork the media 30

31 Gateway selected as recording source Recording Enabled Gateway Gateway Preferred Recording TDMgateway/ CUBE IP phone Recorder [1] [3]: An external call is answered by a User with a Cisco IP phone [4]: The recording session is automatically started from IP phone; the recording media source is Gateway Preferred and is valid for the call flow; Gateway is selected to fork the media 31

32 Alternate source selected Gateway NOT enabled for recording Gateway Preferred Recording TDMgateway/ CUBE IP phone Recorder [1] [3]: An external call is answered by User with IP phone [4]: The recording session is automatically started from IP phone; although the recording media source is Gateway Preferred, the gateway is not enabled for recording; the IP phone is selected to fork the media 32

33 Alternate source selected Both phones can record the same call Gateway Preferred Recording Gateway Preferred Recording calling IP phone called IP phone Recorder [1] [3]: An IP phone User calls another IP phone User [4] - [5]: Two recording sessions are started automatically from both IP phones; the recording media source for both phones is Gateway Preferred, but the gateway is not in the call flow so the IP Phones are selected to fork the media 33

34 Recording session stops when call is held Recording Enabled Gateway Gateway Preferred Recording V TDMgateway/ CUBE 3. MOH 2. User 1 1. User presses hold 4. Recorder [1] - [3]: Call is connected and a gateway recording session has already started. User 1 puts the call on hold, Unified CM plays Music-on-Hold to the caller [4]: Unified CM instructs the gateway to stop forking media to the recorder 34

35 Recording session re-starts when call is resumed Recording Enabled Gateway Gateway Preferred Recording V TDMgateway/ CUBE 6. MOH User 1 5. User presses resume 9. Recorder [5] [8]: User 1 resumes the call, Unified CM removes Music-on-Hold and reconnects the Caller with User 1 [9]: Unified CM instructs the gateway to re-start the recording session 35

36 Recording re-starts when call is resumed from shared line on different device Recording Enabled Gateway Gateway Preferred Recording V TDMgateway/ CUBE 6. User 1 9. MOH Recorder User 2 5. User 2 presses resume [5] [8]: User 2 resumes the call from a shared line on a different device; Unified CM removes Music-on-Hold and connects the caller with User 2 [9]: Unified CM instructs the gateway to re-start the recording session Gateway Preferred Recording 36

37 Recording Call Flows Single Cluster Mobility with mid-call features Remote Destination Profile Jabber for iphone/ipad/android Mobile Voice Access Enterprise Feature Access Automatic recording mode is typically used for illustration. Selective recording modes are supported unless otherwise noted. Mobility feature details: E0EFA0_00_cucm-features-services-guide-91_chapter_ html

38 Recording Mobility Calls Recording Enabled Gateway Gateway Preferred Recording TDMgateway/ CUBE V Mobile Jabber Dual-Mode phone in Wifi mode 4. Recorder [1] - [3]: An external call is answered on a Mobile phone configured as a Dual-mode device [4]: The recording session is automatically started for the dual mobile phone; the recording media source is Gateway Preferred and is valid for the call flow; the Gateway is selected to fork the media 38

39 Selective Recording calls via DTMF Recording Enabled Gateway Gateway Preferred Recording TDMgateway/ CUBE V V User dials *86 6. Recorder [1] [4]: An external call is answered on a remote destination configured on a Remote Destination Profile [5] [6]: The recording session is started by mobile User with DTMF; the ingress gateway is selected to fork the media 39

40 Moving calls between IP Phones and Mobile Phones Recording media source changes mid-call 1. Recording Enabled Gateway V 6. User dials *78 GW Recorder 9. Gateway Preferred Recording [1] [4]: An internal call is extended to a mobile remote phone. [5]: The recording session is started automatically and the egress gateway is selected to fork the media. [6] [8]: Mobile User moves the call to their desktop phone with DTMF. User answers call on their desktop IP Phone. [9]: The gateway is no longer valid for call flow, so the recording session is automatically restarted. The Phone is selected to fork the media. 40

41 Moving calls between IP Phones and Mobile Phones Recording media source changes mid-call Recording Enabled Gateway GW1 1. V Phone Preferred Recording User presses Mobility 7. Recorder 3. [1] [3]: An external User makes a call to an IP phone. IP phone User answers the call. [4] : A recording session is automatically started for the IP Phone. The recording media source is Phone preferred; the phone is selected to fork the media. [5] [6]: IP phone User presses the Mobility softkey to move the call. User answers the call from the mobile phone. [7]: The recording restarts automatically for the remote destination. The recording media source is Gateway preferred; the ingress gateway (GW1) is selected to fork the media Gateway Preferred Recording V GW

42 Recording session stops when call is Held from Mobile phone Recording Enabled Gateway Gateway Preferred Recording GW V RD 3. IP phone User User dials *81 to hold the call Recorder [1] [2]: Mobile User dials DTMF *81 to put the call on hold. [3]: Unified CM plays MOH to the IP phone User [4]: The gateway recording session that already started from the mobile remote destination is stopped 42

43 Recording session re-starts when call is Resumed from Mobile phone Recording Enabled Gateway Gateway Preferred Recording GW V RD 6. IP phone User User dials *83 to resume the call Recorder [5]: Mobile User dials DTMF *83 to resume the call. [6]: Unified CM stops playing MOH to the IP phone [7]: Unified CM reconnects the mobile User to the IP phone User [8]: The recording session from the mobile remote destination is automatically restarted 43

44 Recording in Multi-Cluster Environments Phones can be connected to any cluster Gateways can be connected to any cluster Recorders can be connected to any cluster Recording requests can be exchanged between clusters 44

45 Multi-Cluster Recording SIP Trunk Configuration Cluster 1 Recorder 1 Configure Unified CM SIP trunks to connect to recording-enabled gateways or to forward recording requests to other clusters Cluster 2 45

46 Recording requests can be exchanged across clusters An external call is received on a gateway in Cluster A and is answered on IP phone in Cluster B. Recording is automatically started for IP Phone on Cluster B. The ingress Gateway on Cluster 1 is selected to fork the media. Cluster B signals Cluster A to start the recording. Cluster A instructs the gateway to fork the media to the recorder. Recorder Cluster A 4 Cluster B Called PSTN 2 DATA CENTER LOCAL OFFICE 1 Calling Gateway Preferred Recording Recording is captured 46

47 Multi-site internal company calls may also use gateway recording Multi-site internal company calls may also use gateway recording If Calling party starts recording, GW1 is selected as recording media source If Called party starts recording, GW2 is selected as recording media source. The preferred recording media source may be configured differently on each device (if desired) Calling 1 Gateway preferred Phone preferred 2 Called GW1 GW2 Cluster 1 Cluster 2 3 SAN JOSE, CA Recorder RICHARDSON, TX 4 Recorder 47

48 Network-Recording Use cases IP Phone calls Mobility calls Any mobile device via Single Number Reach (Remote Device Profile) Jabber for Android Jabber for ipad Jabber for iphone Enterprise Feature Access, DTMF Feature Access, Dial-via-Office Extend & Connect calls Jabber for Windows Single & Multi-cluster call flows Check out all the use cases for more information 48

49 Recording Meta-Data Call information is sent to the recorder in the SIP header with each media stream Additional call specific information is retrieved via CTI using the callid 49

50 Recording Meta-Data Available in Unified CM 9.x (or earlier) Data Element Call Identifier - Near-end party Call Identifier - Far-end party Device Name - Near-end party Device Name - Far-end party Called Number - Near-end party Calling Number - Far-end party TAPI Meta-data Request (linecallinfo::devspecific) CCMCallIDInfo.CCMCallId CCMCallIDInfo.CCMCallId linedevcaps.devicename linedevcaps.devicename linedevcaps.dn linedevcaps.dn JTAPI Meta-data Request connection.getconnectionid().i ntvalue() connection.getconnectionid().i ntvalue() terminalconnection.getterminal Name() terminalconnection.getterminal Name() terminalconnection.getaddress () terminalconnection.getaddress () SIP INVITE x-refci x-farendrefci x- nearenddevice x-farenddevice x-nearendaddr x-farendaddr 50

51 Recording Meta-Data New in Unified CM 10.0(1) Data Element TAPI Meta-data Request (linecallinfo::devspecific) JTAPI Meta-data Request SIP INVITE Cluster Identifier - Near-end party linedevcaps.clusterid CiscoProvider.getClusterI D() x- nearendclusterid Cluster Identifier - Far-end party Not applicable Not applicable x-farendclusterid Gateway Call Identifier - Near-end (available only when near-end is Gateway recording media source) RecordingAttributeInfo_ExtD0.GatewayCallProtocolReferen ce terminalconnection.getci scorecorderinfo().getprot ocolreferenceguid() x-nearendguid Gateway Call Identifier - Far-end (available only when far-end is Gateway recording media source) RecordingAttributeInfo_ExtD0.GatewayCallProtocolReferen ce terminalconnection.getci scorecorderinfo().getprot ocolreferenceguid() x-farendguid 51

52 Recording Meta-Data available via CTI-only Data Element TAPI Meta-data Request (linecallinfo::devspecific) JTAPI Meta-data Request SIP INVITE Recorder Profile Directory Number CallAttrtibuteInfo.PartyDN terminalconnection.getciscorecorderinfo().getaddress() N/A Recorder Profile Partition CallAttrtibuteInfo.PartyPartition terminalconnection.getciscorecorderinfo().getaddress() N/A Recorder Sip Trunk Device Name CallAttrtibuteInfo.DeviceName terminalconnection.getciscorecorderinfo().getterminalname() N/A Partition - Near end Partition - Far end linecallinfo.partition terminalconnection.getaddress() N/A linecallinfo.partition terminalconnection.getaddress() N/A Recording Media Source Device Type RecordingAttributeInfo_ExtD0.Fork ingdevicetype terminalconnection.getciscorecorderinfo().getmediaforkingdevicetype() N/A Recording Media Source Cluster Name RecordingAttributeInfo_ExtD0.Fork ingclustername terminalconnection.getciscorecorderinfo().getmediaforkingclusterid() N/A Recording Media Source Device Name RecordingAttributeInfo_ExtD0.Fork ingdevicename 52 terminalconnection.getciscorecorderinfo().getmediaforkingdevicename() N/A

53 SIP Header Recording Meta-Data SIP INVITE to RECORDER: From: x-nearend; x-refci=ci2; x-nearendclusterid=cluster1; x-nearenddevice=sep ; x-nearendaddr=1000 x-farendrefci=ci1; x-farendclusterid=cluster1; x-farenddevice=sipt1gw; x-farendaddr= ; x- farendguid=1f A573F590A> Called 1000 Unified CM Recorder SIPT1GW Signaling Called stream Calling stream PSTN Directory Number: 1000 Device Name: SEP Calling

54 Recording Serviceability Performance Counters monitor real-time and historical feature status Alarms alert administrators when threshold conditions are met 54

55 Recording Serviceability Real-time and Historical Performance Counters available in release 10 Performance Counter Definition Type Recording Gateways In Service number of active/successful registrations to recording-enabled gateways Real-time Recording Gateways Out of Service Recording Gateway Registration failures Gateway Recording sessions active Gateway Recording session failures Phone Recording sessions active number of configured recording gateways with no active registration number of times registration to recording-enabled gateways has failed number of concurrent recording sessions (two streams) from gateway to recorder number of times recording session (two streams) from gateway to recorder has failed number of concurrent recording sessions (two streams) from phone to recorder Real-time Cumulative, Historical Real-time Cumulative, Historical Real-time Phone Recording session failures number of times recording session (two streams) from phone to recorder has failed 55 Cumulative, Historical

56 Recording Alarms available in release 10 Alarm Description Severity Recording Gateway Registration Rejected Recording Gateway Registration Timeout Registration to recording-enabled gateway rejected after multiple attempts; gateway marked out-of-service No response from recording-enabled gateway after multiple attempts; timeout occurred; gateway marked out-of-service Recording Gateway Out Of Service Recording-enabled gateway closed connection to Unified CM Notice Error Error Recording Gateway In Service Recording gateway status changed from out-of-service to inservice Notice Recording Resources Not Available Recording media resources not available (Phone or Gateway) Warning Recording Gateway Session Failed Recording Session Terminated Unexpectedly Recording Call Setup Failed Recording gateway closed the recording session unexpectedly SIP Trunk to the recording server is out-of-service Recording server ends session unexpectedly Call flow not supported Recording call setup failed Recording request to other cluster timed out Error Error Error Recording Invalid Call State Invalid Call State; internal error Info Recording Already 2014 Cisco In and/or Progress its affiliates. All rights reserved. Cisco Recording Public session 56 already in progress Info

57 Recording Server Redundancy 3 Options Redundancy can be configured using a Unified CM Route List and/or DNS SRV Records Load-balancing is vendor dependent Most vendors have implemented a SIP Proxy to provide this service. Use a Route List with two or more SIP Trunks to provide multiple SIP Proxy and Recording server redundancy Option 1 - SIP Redirect Approach: Recorder or SIP Proxy can issue a SIP 3XX response to redirect the Cisco Unified Communications Manager INVITE to load-balance multiple recorders Option 2 - Route List Approach: Multiple SIP Trunks assigned to a Route Group assigned to a Route List - Configure SIP trunk for each addressable Recorder Proxy or Recorder - SIP Trunks are assigned to Recorder Route Group with an assigned algorithm (top-down or circular) - Recorder Route Group assigned to Recorder Route List - Route pattern (e.g. 9XXX) directs traffic to Recorder Route List - Approach provides failover capability and/or load-balancing 57

58 Recording Server Redundancy 3 Options Option 3 - DNS SRV Approach: The SIP Trunk destination address can be populated using a DNS SRV record tag (service, protocol, domain name) DNS SRV records contain: service, protocol, domain name, TTL, class, priority, weight, port and target (Hostname or IP address eg. the recorder or recorder proxy) Multiple SRV records may map to the same SRV record tag (service, protocol, domain name) allowing multiple records to be returned in response to a single DNS SRV query providing redundancy Multiple SRV records can establish the primary and backup targets (different priorities), how to load-balancing across targets (same priority different weights) or a combination of both Redundancy options vary by recording vendor. Please ask your recording vendor which options are supported. 58

59 Call Detail Records Each call recording session generates 2 CDRs; One per media stream (Called + Calling) To identify recording CDRs: Recording CDRs will make use of the onbehalfof field indicating the calls were redirected by the recording feature The Global Call ID fields in the recording CDR will be the same as the call that was recorded 59

60 Interactions & Limitations Multiple recording sessions for the same call is supported when recording is started for both Calling and Called parties Cisco Customer Voice Portal calls may be recorded using Phone as recording media source Gateway may not be valid for some call flows between two Unified CM devices Phone may be selected as the recording media source. Check the recording use cases for more information. SIP Proxy servers may not be placed between the Gateways and Unified CM 60

61 Performance & Scalability Each recording session adds 2 calls to the Busy Hour Call Completion (BHCC) rate for both Unified CM and Gateway Additional impact on CTI resources is minimal 61

62 Recording Details Setting it up..

63 Gateway Requirements Supports both Voice gateways and Unified Border Elements (CUBE) as long as they interface with Unified CM using SIP and the Router platform supports the UC Services Interface (not supported for H323 or MGCP based calls) The word gateway is used interchangeably to refer to both Voice gateways and CUBE devices. The Gateway has to be directly connected to Unified CM using a SIP trunk. ISR-G2 Gateways (29XX, 39XX Series) running release 15.3(3)M1 or later are supported. 15.3(3)M1 was released on CCO in Oct / 2013 ASR-1K Gateways running release XE or later are supported. XE was released on CCO in Oct / 2013 VG224 is not currently supported Check the latest Gateway Requirements here: 63

64 Supported Devices Recording Most Cisco IP Phones support Recording, including Mobility devices and more. For a complete list of devices which support recording, refer to this wiki: 64

65 Provisioning a Recorder 1. Create a new Recording Profile from the Device Settings Menu Assign a DN for the recorder in the Recording Destination Address Configure the Recording Calling Search Space with a partition for the Recorder 2. Create a new SIP Trunk device from the new device configuration page Enter the Device Name and IP address of the recorder Select a partition that will be used for recording Set the Recording Information 3. Create a new Recorder Route Group that contains the Recorder SIP Trunks 4. Create a new Route List that contains the Recorder Route Group 5. Create a new Route Pattern based on the DN for the Recorder and point to Recorder Route List 6. Enable two cluster-wide Recording Tone service parameters (optional): Play to Observed Target (True/False), Play to Connected Target (True/False) 7. Complete the vendor specific guidelines for CTI connections (optional) 65

66 Provisioning a Phone for recording 1. Choose the desired recording option for each line appearance - Automatic Recording or Selective Recording, or Recording Disabled (default) 2. Select a Recording Profile for each line appearance 3. Set the Recording Media Source preference Phone preferred or Gateway preferred 4. If Phone recording is desired, enable the Built-In-Bridge (BIB) on the Device page 5. To enable Selective User recording, add the record softkey or programmable line key to the device template and associate it with the IP phone 6. Disable codecs (e.g. G.722/iSAC/iLBC) if the Recorder does not support them using either the available codec Service Parameters or the Audio Codec Preference List 66

67 Provisioning Mobility DTMF for recording *86 DTMF sequence may be used to start/stop a recording session from Cisco Mobility devices Set the Enable Enterprise Feature Access Service Parameter to True. 67

68 Provisioning Mobility DTMF for recording Set the DTMF Signaling Method configured on the Trunk to match the configuration of the physical trunk. 68

69 Setting up a Gateway for Recording The following example sets up the router for Cisco Unified Communication IOS Services. It enables the HTTP server and the XMF, providers. The configuration specifies the address and port that the application uses to communicate with the XMF provider.! uc wsapi message-exchange max-failures 100 response-timeout 0 source-address probing interval negative 20 probing interval keepalive 255 probing max-failures 3! provider xmf remote-url 1 remote-url 2 CLI s to enable UC Services API CLI s to enable HTTP Server! http client connection timeout <1-60> http client connection idle timeout <1-600>! ip http server ip http timeout-policy idle 600 life requests ip http max-connections 1000 Here 2 XMF applications are configured with the applications ID s 1 (for Subscriber 1) and 2 (for Subscriber 2). Up to 32 applications can be configured. 69

70 Setting up a Gateway for Recording! Sample router configuration! Timeouts shown here are typical values! Depending on customer environment values may need to be tuned http client connection timeout 60 http client connection idle timeout 600 ip http server ip http max-connections 100 ip http timeout-policy idle 600 life requests uc wsapi message-exchange max-failures 3 source-address r6h24-cube1.cisco.com probing interval negative 10 probing interval keepalive 180! provider xmf remote-url 1 UCM Publisher remote-url 2 UCM Subscriber 70

71 Setting up a Gateway for Recording! Common show and debug commands! Show commands show wsapi registration xmf! List of Registered UCM! Debug commands debug voip application! App framework debug debug voip application media forking! RTP Forking info debug ccsip message! SIP Signaling debug wsapi xmf message! XMF signaling debug voip rtp packet! RTP packet flow Refer to Cisco IOS WSAPI reference for complete details: 71

72 When configuring the feature using IOS 15.4(1)T1 or XE3.12., refer to this guide: 3s/cube-proto-xe-3s-book/voi-cube-uc-gateway-services.html Here are the minimum steps required: 1. Enable HTTP on IOS ip http server http client persistent 2. Enable the API on IOS uc wsapi source-address [IP_Address_of_the_Router] 3. Enable XMF service within the API provider xmf remote-url 1 CM Subscriber IP Address:8090/xmf1 no shutdown

73 Usage & Integration

74 Assign different recorders to each recording media source Multi-cluster, multi-site Customers may want to use a different recording server based on the recording media source selected When gateway is selected, use Central recorder When phone is selected, use Branch recorder Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Phone Preferred Recording ISR- G2/CUBE V External call Called Internal call Calling central recorder branch recorder 74

75 Assign different recorders to each recording media source For this example, let s assume: Route pattern for Central Recorder is 1111 Route pattern for Branch Recorder is 2222 Add a Translation Pattern on Cluster 2. The translation pattern will transform the number 1111 to Assign the translation pattern to a special partition, say recorder_partition. Create a Calling Search Space on Cluster 2, say recorder_css, to include recorder_partition. Configure the recording profile for the called party line appearance on Cluster 2 to use the Central Recorder on cluster 1, but set the calling search space of the recording profile to recorder_css. When the gateway is selected, the 1111 of the recording profile is sent to Cluster 1, so the Central Recorder receives the media. The recording calling search space "recorder_css" is not sent to or used in cluster 1 since calling search spaces are cluster-specific. When the phone is selected, cluster 2 will search for 1111 using recorder_css which matches the translation pattern The number gets translated to 2222 and 2222 is the route pattern for the SIP trunk pointing to the Branch Recorder. 75

76 Assign different recorders to each recording media source Cluster 2 Gateway selected Recorder server address: 1111 sent to Cluster 1 Recording css: recorder_css (not applicable) Cluster 1 Route patter: > SIP Trunk to Central Recorder Phone selected Recorder server address: 1111 (processed locally) recorder_css: recorder_css translation pattern 1111 (recorder_partition), transforms to 2222 route pattern (2222) -->SIP Trunk to Branch Recorder 76

77 Contact Center Integration Enterprise/Hosted (and later) Requires Cisco Unified CM 6.0(1) or later Cisco Agent Desktop 7.2(1) uses its own desktop, pc-based Recording mechanism prior to UCCE 8.0 Cisco Agent Desktop 8.0 supports interoperability with Unified CM Recording CTI Toolkit supports Unified CM Recording Finesse supports Unified CM Recording For all desktops, when using call recording feature the 3rd-party recording server will monitor the UCCE CTI server. When the Agent/Supervisor presses the Start/Stop Call Record buttons on the desktop, the 3rd-party recording server sees those events and in turn, instructs Unified CM to begin recording the call using the CTI interface. 77

78 Contact Center Integration Unified Contact Center Express CAD Agent/Supervisor clients use their own built-in call recording mechanism Finesse Agent/Supervisor clients support Unified CM Recording Mobile Agent solution using Extend & Connect supports Unified CM Recording feature in release 10 or later Genesys Contact Center T-Server 8.0 (and later) supports Unified CM Recording 78

79 Cisco MediaSense Recording & Playback Full-time and On-Demand recording Centralized and Branch network topology friendly Contact Center Agent & Remote Agent recording Easy Search and Play-back Listen to LIVE calls recorded in real-time Tag recordings for review later Tight integration with Cisco Collaboration Portfolio Adds Video-on-Hold to Unified CM calls Partner application eco-system 79

80 Partner Integrations & Sales Contacts (1 of 3) Contact Partner to determine individual feature support ASC Telecom Americas: Europe: Asia/Pacific: Autonomy/eTalk Tuan Le Calabrio Americas & International: Cistera Networks USA Sales: CTI Group, Inc. - Offers separate silent monitoring application Americas: ahash@ctigroup.com EMEA: tdavis@ctigroup.com Cyber-Tech International - Offers separate silent monitoring application Arno Sybrandy Sales@cybertech-int.com

81 Partner Integrations & Sales Contacts (2 of 3) Contact Partner to determine individual feature support KnoahSoft - Offers separate silent monitoring application Americas & Europe: Walter Kenrich wkenrich@knoahsoft.com Asia & India: Subhash Kothuru subhash.k@knoahsoft.com or NICE Systems - Offers separate silent monitoring application Americas: nice.sales.americas@nice.com Europe: nice.sales.emea@nice.com Asia/Pacific: nice.sales.apac@nice.com TC&C Telecommunication International: sales@tcandc.com USA & Canada: sales-usa@tcandc.com EU Nordic: sales-nordic@tcandc.com Telrex North America: sales@telrex.com x2 Central and Latin America: Diana Oriel doriel@telrex.com x111 Europe and Asia Pacific: Jim Roark jroark@telrex.com x106 telrex com/about_us_contact htm 81

82 Partner Integrations & Sales Contacts (3 of 3) Contact Partner to determine individual feature support Telstrat Bob Dudas Verba - Offers separate silent monitoring application Americas: sales@verba.com VERBA (83722) EMEA/ASIAPAC: sales@verba.com Verint (Witness) Systems Americas: Ken Carney ken.carney@verint.com Europe: John Crosby john.crosby@verint.com + 44 (0) Asia/Pacific: Contact APAC Marketing Team marketing.hk@verint.com Voice Print International Chris Morrissey cmorrissey@vpi-corp.com Search for the latest list of Recording Partners using the Solutions Catalog: 82

83 Possible Failure Scenarios Single-cluster Recording Both the User device and gateway are not capable of media forking Recording setup failure due to unreachable recorder address or wrong Calling Search Space configuration Recording setup failure due to media connection setup failure Gateway media forking failed Conference setup failure, reserved for two-party calls Multi-cluster Recording Recording setup failed due to unreachable recorder Both the User device and gateway are not capable of media forking Gateway media forking failed Conference setup failure reserved for two-party calls 83

84 Secure Recording Introduced in Unified CM 8.0(1)

85 Secure Recording Introduced Unified CM 8.0(1) Maintains call security Recording calls are always established using the highest level of security determined by the capabilities of the device being recorded regardless of the security status of the call being recorded Recording servers must use a Secure SIP Trunk connection to capture media from devices enabled for encrypted media (srtp) When the recording server is not configured to support encrypted media, requests to record secure-enabled devices are rejected Recording calls using secured media carries approximately 4k of additional bandwidth overhead per call; same as standard secure (srtp) calls 85

86 Secure Recording Recording server must meet or exceed security capabilities of the device to capture the encrypted call. Device security capabilities the two recording calls are established at the highest security level permitted by agent device and recorder s security capabilities e e e SIP E E E Calling Called Recorder 86

87 Secure SIP Trunk Provisioning Create a Secure SIP trunk Profile Set the Device Security Mode to Encrypted Allow Unified CM to transmit call security status over the SIP Trunk by enabling the Transmit Security Status checkbox Apply the security profile to the SIP trunk for the recorder. To enable srtp, also enable the Allow srtp checkbox on the SIP Trunk Device page. Unified CM Media Termination Points (MTP) support pass thru codec, but do not support srtp directly, so verify Media Termination Point Required is unchecked when using Unified CM software-based MTPs (e.g. ipvms service). Gateway ios MTPs support srtp and can be configured with and w/o pass thru codec. 87

88 Recording non-secure calls Use case 1 The Called device is non-secure and the Recorder is secure. Since the recording server exceeds the security capabilities of the Called device, Recording is allowed to proceed. The overall security of the call is non-secure. e u SIP e U U U Calling Called Recorder 88

89 Recording non-secure calls Use case 2 The Called device is secure and the Recorder is non-secure. Since the Recorder does not meet the security capabilities of the Called device, the recording calls are disconnected as soon as the call is answered the recorder s security capabilities are only known after the call is connected Cause code 57 (Bearer capability not authorized) will be used to clear these calls. 403 Forbidden is the corresponding SIP event triggered by Q.931 cause code 57 Bearer capability not authorized. CTI will notify application that the Recording session has been disconnected. u e u SIP Calling U Called 89 U U Recorder

90 Secure Media Key Exchange Individual security keys are issued for each stream/call to maintain the highest level of security Overall security status of the call (monitoring and recording session) e e Calling srtp KeyA E e Called (Agent srtp KeyB E srtp KeyC E E Supervisor e srtp KeyD SIP Recorder 90

91 Recording secure and non-secure calls All devices support secure media, so all calls are encrypted e srtp KeyA e srtp KeyB SIP e E E E Calling Called srtp KeyC Recorder Called device and Recorder both support secure media, so the recording session is encrypted, while the actual call remains unencrypted u e srtp KeyA SIP e Calling U Called E E srtp KeyB Recorder 91

92 Authenticated devices are not considered secure The Called device is authenticated, but not enabled for encrypted media. Recording is not supported on this device. Invites are not sent to the Recorder when the Called device is authenticated, but does not support encrypted media. u a No invite is sent to recorder e SIP Calling U Called Recorder e Calling A a Called No invite is sent to recorder SIP u Recorder 92

93 Secure recording of secured call over MTP Cisco Media Termination Points (MTP) do not support encrypted media. If an MTP is inserted in the Secure SIP trunk connecting the Called device to the Recorder, the recording calls are cleared. The same result occurs if the Recorder SIP trunk is non secure. 93

94 Recording secure calls when transcoders are used A transcoder is inserted between the Called device and the Recorder because of codec incompatibility. Cisco transcoders do not support encrypted media, so the Recording calls are disconnected Cause Code 57 (Bearer capability presently not authorized) will be used to clear the call. 403 Forbidden is the corresponding SIP event triggered by Q.931 cause code 57 Bearer capability not authorized. Cisco CTI will notify the application that the Recording session has been disconnected. u e SIP e U U Calling Called Recorder 94

95 Secure Tone Interaction Secure Tone was introduced in Unified CM 7.0(1) to provide call participants an audible indication their call is secured (using encrypted media, srtp) When Secure Tone is enabled, it will play once to call participants at the beginning of the call If Secure Tones and Recording Tones are both enabled, the Secure Tone will play once, followed by Recording Tones (if the call is being recorded) 95

96 Recording continues when customer puts the call on hold and non-secured MOH is inserted to the agent A secure recording session is established. The Calling party places the Called party on hold. If the Calling party is a Unified CM User, Music-On-Hold may be inserted into the media stream. Since MOH is not secure, the overall security status of the call is downgraded. The secure icons of Calling and Called devices are updated. The recording session continues because the Recorder meets or exceeds the security capabilities of the (Called) device being recorded e 1 hold MOH server e SIP e Calling U Called E E Recorder 96

97 Complete Your Online Session Evaluation Give us your feedback and you could win fabulous prizes. Winners announced daily. Complete your session evaluation through the Cisco Live mobile app or visit one of the interactive kiosks located throughout the convention center. Don t forget: Cisco Live sessions will be available for viewing on-demand after the event at CiscoLive.com/Online 97

98 Continue Your Education Check out all of the call flows in the back of this slide deck Demos in the Cisco Campus Walk-in Self-Paced Labs Table Topics Meet the Engineer 1:1 meetings Questions? 98

99

100 Additional Recording Call Flows

101 Recording Call Flows Single Cluster IP Phone Including Mid-Call features: Hold/Resume, Shared-lines Transfer Conference Automatic recording mode is used for illustrations. Selective recording modes are supported unless otherwise noted.

102 External Call to IP phone Selective Recording Gateway preferred, Gateway selected Recording Enabled Gateway Gateway Preferred Recording IP phone TDMgateway/ CUBE User presses Record softkey 6.Phone displays Recording and changes softkey label Recorder [1] [3]: An external call is answered by a User with a Cisco IP phone [4] [5]: IP phone User starts a new recording session by pressing the Record softkey. The recording media source is Gateway Preferred and is valid for the call flow. The gateway is selected to fork the media [6] IP phone displays Recording and the softkey label changes to StopRec 102

103 External Call to Jabber Softphone Selective Recording Gateway preferred, Gateway selected Recording Enabled Gateway Gateway Preferred Recording Jabber Softphone TDMgateway/ CUBE CTI application invokes silent recording Recorder [1] [3]: An external call is answered by User with the Jabber softphone [4] [5]: CTI application starts a selective silent recording session; the recording media source is Gateway Preferred and is valid for the call flow; the Gateway is selected to fork the media 103

104 Mid-Call - External Call Transfer for User 1 to User 2 (pic 1) Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE 3. User 2 Recorder [1] [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1 s IP Phone; the ingress gateway is selected to fork the media 104

105 Mid-Call - External Call Transfer for User 1 to User 2 (pic 2) Transfer Consultation Call Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE 7. MOH press transfer 10.dial User User 2 Recorder Gateway Preferred Recording [5] [7]: User 1 presses transfer softkey, which puts the external call on hold [8]: The recording session for the external call is stopped [9]: MOH is played to the external caller [10] [13]: User 1 makes a consultation call to User 2, User 2 answers the call [14] [15]: The two recording sessions are started for User 1 and the User 2, IP phones are selected to fork the media 105

106 Mid-Call - External Call Transfer for User 1 to User 2 (pic 3) Transfer is completed Recording Enabled Gateway 18. Gateway Preferred Recording TDMgateway/ CUBE 17. MOH User 1 16.presses transfer 22. User 2 Recorder [16]: User 1 presses transfer softkey again to complete the transfer [17] [20]: Unified CM splits the primary call and the consultation call for User 1. [21] [22]: The recording sessions for User 1 and User 2 from the consultation call are stopped [23] [24]: Unified CM joins the gateway side call leg of the primary call with User 2 side call leg of the consultation call [25]: The recording session is re-started for User 2; the gateway is selected to fork the media 106

107 Mid-Call - External Call Transfer from IP Phone to a non-bib Device (pic 1) Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE User 2 (Jabber for ipad) Recorder [1] [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1; the ingress gateway is selected to fork the media 107

108 Mid-Call - External Call Transfer from IP Phone to a non-bib Device (pic 2) Transfer Consultation Call Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE 7. MOH press transfer 10.dial User Recorder User 2 (Jabber for ipad) 15.recording invocation fails [5] [7]: User 1 presses the transfer softkey, which puts the external call on hold [8]: The recording session for the external call is stopped [9]: MOH is played to the external caller [10] [13]: User 1 makes the consultation call to User 2, User 2 answers the call [14]: The recording session for User 1 is started. [15]: The recording session for User 2 fails, since Phone-based media forking is not supported for this device type 108

109 Mid-Call - External Call Transfer from IP Phone to non-bib Device (pic 3) Transfer is completed Recording Enabled Gateway Gateway Preferred Recording 18. TDMgateway/ CUBE 17. MOH User 1 16.press transfer Recorder User 2 (Jabber for ipad) [16]: User 1 presses the transfer softkey again to complete the transfer [17] [20]: Unified CM splits the primary call and the consultation call for User 1, [21]: The recording session for the consultation call is stopped. [22] [23]: Unified CM joins the gateway side call leg of the primary call with User 2 side call leg of the consultation call [24]: The recording session is re-started for User 2; the gateway is selected to fork the media 109

110 Mid-Call - Internal Call Transfer to an External User (pic 1) Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE User 2 Recorder [1] [3]: User 1 calls User 2, User 2 answers [4] [5]: Recording is automatically started for both User 1 and User 2; both Phones are selected to fork the media 110

111 Mid-Call - Internal Call Transfer to an External User (pic 2) Consultation Call Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE MOH 6.press transfer 11.dial User 2 User 2 Recorder [6] [8]: User 1 presses transfer softkey, which puts User 2 on hold [9]: The recording session for User 1 is stopped; the recording session for User 2 continues. [10]: MOH is played to User 2. [11] [13]: User 1 makes a consultation call to an external User [14]: A recording session for the consultation call is established; the gateway is selected to fork the media 111

112 Mid-Call - Internal Call Transfer to an External User (pic 2) Transfer is completed Recording Enabled Gateway 19. Gateway Preferred Recording User 1 TDMgateway/ CUBE MOH presses transfer 18. User 2 Recorder [15]: User 1 presses the transfer softkey again to complete the transfer [16]: Unified CM stops playing MOH to User 2 [17] [18]: Unified CM split the primary call for User 1, the associated recording session for User 2 is stopped [19] [20]: Unified CM split the consultation call, the associated recording session for User 1 is stopped [21] [22]: Unified CM joins the User 2 with external User [23]: The recording session is re-started for User 2; the gateway is selected to fork the media 112

113 Mid-Call - External Call Transfer from IP Phone to another IP Phone Recording media source is dynamically selected Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE transfer call to User Phone Preferred Recording 8. User 2 Recorder [1] [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1; the gateway is selected to fork the media [5] [7]: User 1 transfers the call to User 2 [8]: The recording session is re-started for User 2; the phone is selected to fork the media 113

114 Mid-Call - External Call is Redirected from IP Phone to another IP Phone Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE User 2 Recorder [1] [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1; the ingress gateway is selected to fork the media 114

115 Mid-Call - External Call is Redirected from IP Phone to another IP Phone Redirected call is connected Recording Enabled Gateway Gateway Preferred Recording 6. User 1 TDMgateway/ CUBE redirect the call to User 2 User 2 Recorder [5] [7]: User 1 redirects the external call to User 2 using a CTI application [8]: The recording session for User 1 is stopped once the call is redirected [9] [10]: Unified CM connects the redirected call [11]: Recording session for User 2 is established; the gateway is selected to fork the media 115

116 Mid-Call - External Call Redirected from IP Phone to another IP Phone Recording media source is dynamically selected Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE redirect the call to User Phone Preferred Recording 8. User 2 Recorder [1] [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1; the gateway is selected to fork the media [5] [7]: User 1 redirects the call to User 2 [8]: The recording session is re-started for User 2; the phone is selected to fork the media. The recording session established in step 4 ends when User 1 s call is redirected 116

117 Mid-Call - External Call Redirected from IP Phone to another IP Phone Recording media source is dynamically selected Recording Enabled Gateway Phone Preferred Recording User 1 TDMgateway/ CUBE redirect call to User Gateway Preferred Recording User 2 Recorder [1] [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1; the phone is selected to fork the media [5] [7]: User 1 redirects the call to User 2 [8]: The recording session is started for User 2; the gateway is selected to fork the media. The recording session established in step 4 ends when User 1 s call is redirected 117

118 Mid-Call - External Call Conference from IP Phone to another IP Phone (pic 1) Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE User 2 Recorder [1] - [3]: An external call is answered by User 1 [4]: The recording is automatically started for User 1; the ingress gateway is selected to fork the media 118

119 Mid-Call - External Call Conference from IP Phone to another IP Phone (pic 2) Consultation call Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE 7. MOH press conference 10.dial User User 2 Recorder [5] [7]: User 1 presses the conference softkey, which puts the external call on hold [8]: The recording session for the external call is stopped [9]: MOH is played for the external caller [10] [13]: User 1 makes the consultation call to User 2 [14] [15]: Two recording sessions are started for User 1 and the User 2; both phones are selected to fork the media 119

120 Mid-Call - External Call Conference from IP Phone to another IP Phone (pic 3) Completing the conference Recording Enabled Gateway Gateway Preferred Recording User 1 TDMgateway/ CUBE MOH CFB User presses conference Recorder [16]: User 1 press the conference softkey again to complete the conference [17] [20]: Unified CM splits the primary call and the consultation call for User 1 [21] [22]: The two recording sessions for the consultation call are stopped [23] [25]: Unified CM redirects User 1 to the conference bridge. A new recording session is started for User 1; the phone is selected to fork the media [26] [27]: Unified CM redirects the external User to the conference bridge, [28] [30]: Unified CM redirects User 2 to conference bridge. A new recording session is started for User 2, the phone is selected to fork the media User 2

121 Recorder redirects recording to new destination Two Recorders are registered with a SIP proxy Cisco Unified CM sends two call setup messages to the BiB (Called) device 1st call is for Called Party stream 2nd call is for Calling Party stream 1 dial setup 4 setup answer INVITE 6 INVITE Recorder 1 DN=3000 Unified CM INVITES Recorder 1 to both calls Recorder returns 3xx with Recorder 2 DN in response to the INVITEs Calling Called Recorder 2 DN=3001 Unified CM redirects the recording call by sending new Setup messages to the Called Party phone and new INVITEs to Recorder 2 9 INVITE 10 INVITE 7 3XX 8 3XX Recorder 1 DN=3000 Called party device sends both media streams to Recorder 2 Recorder 2 may choose to reply with another Redirect 3XX message Unified CM will process the Redirect accordingly Calling Called Recorder 2 DN=3001 Calling Called 121

122 Simultaneous monitoring and recording Each phone can simultaneously support 1 monitoring and 1 recording session Recorder Supervisor Desktop agent 1 on call 1 [monitor] agent 2 idle [monitor ] agent 3 idle [monitor ] 5 4 setup 3 setup 1 dial 2 answer 7 (accept) 6 dial(monitor) Calling Called (Agent) Supervisor Calling voice Called (Agent) voice mix of Agent + Calling party voice 122

123 Simultaneous monitoring and recording of Observer A monitoring end-point can also be monitored or recorded by another end-point Supervisor Desktop agent 1 on call1 [monitor] agent 2 idle [monitor ] agent 3 idle [monitor ] 3 2 nd level Supervisor Desktop supervisor on call 1 [monitor] agent 1 idle [monitor ] agent 2 idle [monitor ] 8 The diagram shows Supervisor 2 Recording and Monitoring Supervisor 1 while Supervisor 1 is monitoring Agent 1 Recorder 7 setup The monitoring stream to Supervisor 2 is the mixture of the local stream (Supervisor 1 s voice) and the remote stream (the mix of the Caller and Agent 1) The recording session for Supervisor 1 only contains a single stream of Agent and Caller s voices 1 Calling dial 2 answer Calling voice Agent voice Agent 5 (accept) 4 dial(monitor) 10 Supervisor 1 DN=1000 (accept) mix of Agent + Calling voice 6 setup dial(monitor) 2 nd Level Supervisor 2 DN=1001 mix of Agent + Calling + Supervisor 1's voice 9 123

124 Recording Call Flows Single Cluster Mobility with mid-call features Remote Destination Profile Jabber for iphone/ipad/android Mobile Voice Access Enterprise Feature Access Automatic recording mode is typically used for illustration. Selective recording modes are supported unless otherwise noted. Mobility feature details: 0_00_cucm-features-services-guide-91_chapter_ html

125 Mobility - Jabber Softphone to Remote Destination Profile Gateway not valid for Jabber call flow Gateway Preferred Recording Gateway Preferred Recording Recording Enabled Gateway Jabber softphone V User presses *86 [1] [4]: An internal call is answered on a remote destination configured on a Remote Destination Profile [5]: Recording is started from Jabber softphone. From Jabber perspective, the call is answered on the Remote Destination Profile, so the gateway is not considered to be in the call flow; the softphone is selected to fork the media [6] [7]: A second recording session is started by mobile User with DTMF, the egress gateway is selected to fork the media Recorder 125

126 Mobility - Mobile DVOR Call to a IP Phone Recording Enabled Gateway Gateway Preferred Recording Gateway Preferred Recording GW V User press * User 7. press Record softkey Recorder [1]: Mobile User 1 calls an internal IP phone User via DVOR, an SIP Invite sent to Unified CM via data channel. [2] - [3]: Unified CM makes an outbound call to the mobile phone [4] - [5]: Mobile User is connected to the IP phone User [6] [7]: IP phone User starts recording with softkey. IP Phone source is Gateway Preferred, but the call is connected to another cluster device. From the Phone perspective the gateway is not in the call flow so, the Phone is selected to fork the media [8] [9]: Mobile User starts recording with DTMF; the Gateway is selected to fork the media

127 Mobility - Mobile DVOF to Remote Destination Profile Gateway Preferred Recording Gateway Preferred Recording Gateway recording NOT enabled GW1 V V GW USER 1 7. User press *86 Recorder USER 2 6. User presses *86 but recording fails [1] [5]: User 1 makes a DVOF call to User 2 who answers on a remote destination configured on a Remote Destination Profile [6]: User 2 attempts to start recording with DTMF. The recording fails since gateway (GW2) is the only gateway in the call flow for User 2 perspective and GW2 is NOT enabled for recording. [7] [8]: User 1 successfully starts a recording session. Gateway (GW1) is selected to fork the media 127

128 Mobility - Mobile Phone Inbound Call with Mobile Voice Access MVA Service hosted in Unified CM is running on Gateway MVA V 2. Gateway Preferred Recording GW Recorder [1]: Mobile phone makes MVA call. First the call is connected to the MVA service at the gateway. User is prompted for PIN and destination number. After digits are collected, the MVA service delivers the call to Unified CM. [3] [5]: Unified CM setups the call which is answered on IP Phone. [6]: Recording is automatically started for mobile phone after the call is active. The gateway is selected to fork the media. 128

129 Mobility - Mobile Phone Inbound Call with Enterprise Feature Access Gateway Preferred Recording V IMS IMS feature handling the EFA call GW User enter PIN, and then target DN 7. User enter *86 to start recording Recorder [1] [3]: Mobile phone dials the EFA number configured in Unified CM. The call is answered by the IMS feature. [4]: Mobile User enters the PIN number and the desired destination DN. [5] [6]: Unified CM setups the call which is answered on Cisco IP Phone. [7] [8]: Mobile User uses DTMF to start the recording session. The gateway is selected to fork the media. 129

130 Mid-Call - External Call Transfer from Mobile Remote Destination to another IP Phone (pic 1) Mobile Transfer Consultation Call Recording Enabled Gateway Gateway Preferred Recording V GW1 MOH GW enter *82 to put the call 11. hold 6. mobile User dials EFA IP phone User DID, then enter Recorder <PIN>#*84#<IP phone User DN>#*84 [1] [3]: The mobile User dials DTMF *82 to put the active call with the external caller on hold [4]: Unified CM plays MOH to the external caller [5]: The recording session for the external call is stopped [6] [9]: The mobile User makes a new call to EFA DID, and enters <PIN>#*84#<IP phone User DN>#*84, the IP phone User answers [10]: The recording session from the mobile remote destination is started; the egress gateway is selected to fork the media [11]: The recording session from IP phone is started; the phone is selected to fork the media 130

131 Mid-Call - External Call Transfer from Mobile Remote Destination to another IP Phone (pic 2) Mobile Transfer Completes Recording Enabled Gateway Gateway Preferred Recording 13. GW1 18. MOH GW2 V Recorder 17. IP phone User 12. enter *84 to complete the transfer [12]: The mobile User dials DTMF *84 to complete the transfer [13] [15]: Unified CM splits the primary call and the consultation call from mobile remote destination [16] [17]: Unified CM stops the recording sessions from mobile remote destination and the IP phone User [18] [20]: Unified CM joins the external caller call leg with the IP phone User call leg [21]: The recording session from IP phone is re-started; the gateway is selected to fork the media 131

132 Mid-Call - External Call Conference from Mobile Remote Destination to IP Phone (pic 1) Mobile Consultation call Recording Enabled Gateway TDMgateway/ CUBE MOH 2. Gateway Preferred Recording V Recorder [1] [3]: The mobile User dials DTMF *82 to put the current call on hold [4]: Unified CM plays MOH to the external caller [5]: The recording session for the external call is stopped [6] [9]: The mobile User makes a new call to EFA DID, and enters <PIN>#*85#<IP phone User DN>#*85#, the IP phone User answers [10] [11]: Two recording sessions are started from mobile User and the IP phone User. The gateway is selected to fork the media for the Mobile User. The Phone is selected to fork the media for the IP Phone User IP phone User 1. enter *82 to put the call hold 6. The mobile User User 1 makes new call to EFA DID, and then enter <PIN>#*85#<IP phone User DN>#*85# 132

133 Mid-Call - External Call Conference from Mobile Remote Destination to IP Phone (pic 2) Mobile Conference Completes Recording Enabled Gateway Gateway Preferred Recording V V TDMgateway/ CUBE 22. MOH CFB [12]: The mobile User dials DTMF *85 to complete the conference [13] [16]: Unified CM splits the primary call and the consultation call from the mobile User [17] [18]: The two recording sessions for the consultation call are stopped [19] [24]: Unified CM redirects the mobile User, the external caller, and the IP phone User to the conference bridge [25]: The recording session for the mobile User is (re)started with the egress gateway selected to fork the media. [26]: The recording session for the IP phone User is started with the Phone selected to fork the media Recorder IP phone User 12. enter *85 to complete the conference

134 Recording Call Flows Single Cluster Extend & Connect CTI Remote Device & Cisco Jabber Automatic recording mode is typically used for illustration. Selective recording modes are supported unless otherwise noted. Extend & Connect feature details: _C3E0EFA0_00_cucm-features-services-guide-91_chapter_ html

135 Extend & Connect - External Call to CTI Remote Device Gateway preferred, Gateway selected Recording Enabled Gateway Gateway Preferred Recording V CTI Remote Device 3. GW V TDMgateway/ CUBE Recorder [1] [4]: An external call is answered on a Extend & Connect remote destination configured as a CTI Remote Device [5]: The recording session is started from the CTI Remote Device; the recording media source is Gateway Preferred and is valid for the call flow; the ingress gateway is the first gateway in the call flow, so the ingress gateway is selected to fork the media 135

136 Extend & Connect - External Call to CTI Remote Device Alternate Gateway Selected Gateway Recording NOT enabled GW V Gateway Preferred Recording CTI Remote Device 4. Recording Enabled Gateway 3. GW V Recorder [1] [4]: An external call is answered on a Extend & Connect remote destination configured as a CTI Remote Device [5]: The recording session is started from the CTI Remote Device; the recording media source is Gateway Preferred and is valid for the call flow; the ingress gateway is not enabled for recording, so the egress gateway is selected to fork the media 136

137 Extend & Connect - IP Phone to CTI Remote Device Gateway not valid for IP Phone IP phone Gateway Preferred Recording Gateway Preferred Recording CTI Remote Device 3. Recording Enabled Gateway GW V Recorder [1] [4]: An external call is answered on a Extend & Connect remote destination configured as a CTI Remote Device [5]: Two recording sessions are started for each device. IP Phone is Gateway Preferred, but the call is connected to another User device. From the IP Phone perspective, the gateway is not in the call flow so the Phone is selected to fork the media [6]: CTI Remote Device is Gateway Preferred, so the egress gateway is selected (which is the first gateway in the call flow). 137

138 Extend & Connect - External Call to CTI Remote Device DTMF not supported for device Recording Enabled Gateway Gateway Preferred Recording V CTI Remote Device 3. GW V TDMgateway/ CUBE User presses*86 but recording invocation fails [1] [4]: An external call is answered on a Extend & Connect remote destination configured as a CTI Remote Device [5]: The mobile User attempts to start a selective recording session with DTMF but it fails; DTMF recording controls are not supported for this device type. Refer to recording mode compatibility matrix: /wiki/main/unified+cm+silent+monitoring+recording+supported+device+matrix Recorder 138

139 Recording Call Flows Multi-Cluster IP Phone Including Mid-Call features: Hold/Resume, Shared-lines Transfer Conference Automatic recording mode is used for illustrations. Selective recording modes are supported unless otherwise noted.

140 Inter-cluster Recording IP Phone call from Cluster 1 to IP Phone on Cluster 2 Phone preferred, Phone selected Phone preferred recording Cluster 1 Cluster 2 SIP Trunk ICT Phone preferred recording IP phone IP phone central recorder branch recorder [1] [4] An IP phone in Cluster 1 calls another IP phone in Cluster 2. [5]: Recording is automatically started for IP Phone on Cluster 2; the Phone is selected to fork the media. The recording profile assigned to the Phone is configured to use the branch recorder. [6]: Recording is automatically started for IP Phone on Cluster 1; the Phone is selected to fork the media. The recording profile assigned to the Phone is configured to use the central recorder. 140

141 Inter-cluster Recording External call from Cluster 1 to IP Phone on Cluster 2 Gateway selected, Recording Profile specifies Central recorder Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Gateway Preferred Recording ISR- G2/CUBE V IP phone central recorder branch recorder [1] [4] An external call is received on a gateway in Cluster 1 and is answered on IP phone in Cluster 2. [5]: Recording is automatically started for IP Phone on Cluster 2. The ingress Gateway is selected to fork the media. Cluster 2 signals Cluster 1 to start the recording. [6]: Cluster 1 instructs the gateway to fork the media to the central recorder. The recording profile assigned to IP Phone in Cluster 2 is configured to use the central recorder. 141

142 Inter-cluster Recording - External call from Cluster 1 to IP Phone on Cluster 2 Phone selected, Recording Profile specifies Branch recorder Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Phone Preferred Recording V ISR- G2/CUBE IP phone central recorder branch recorder [1] [4] An external call is received on a gateway in Cluster 1 and is answered on IP phone in Cluster 2. [5]: Recording is automatically started for IP Phone on Cluster 2; the Phone is selected to fork the media. The recording profile assigned to the IP Phone is configured to use the branch recorder. 142

143 Inter-cluster Recording - External call from Cluster 1 to IP Phone on Cluster 2 Gateway selected, Recording Profile specifies Branch recorder Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Gateway Preferred Recording V ISR- G2/CUBE IP phone central recorder branch recorder [1] [4] An external call is received on a gateway in Cluster 1 and is answered on IP phone in Cluster 2. [5]: Recording is automatically started for IP Phone on Cluster 2. The ingress Gateway is selected to fork the media. Cluster 2 signals Cluster 1 to start the recording. [6]: Cluster 1 instructs the gateway to fork the media to the branch recorder. The recording profile assigned to IP Phone in Cluster 2 is configured to use the branch recorder. 143

144 Inter-cluster Recording - External call from Cluster 1 to IP Phone on Cluster 2 Phone selected, Recording Profile specifies Central recorder Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Phone Preferred Recording V ISR- G2/CUBE IP phone central recorder branch recorder [1] [4] An external call is received on a gateway in Cluster 1 and is answered on IP phone in Cluster 2. [5]: Recording is automatically started for IP Phone on Cluster 2; the Phone is selected to fork the media. The recording profile assigned to the IP Phone is configured to use the central recorder. 144

145 Network-based Recording Call Flows Multi-Cluster Mobility Remote Destination Profile Jabber for iphone/ipad/android Mobile Voice Access Enterprise Feature Access Automatic recording mode is typically used for illustration. Selective recording modes are supported unless otherwise noted. Mobility feature details: C3E0EFA0_00_cucm-features-services-guide-91_chapter_ html 145

146 Inter-cluster Recording IP Phone call from Cluster 2 to Remote Device Profile on Cluster 1 Gateway selected, Recording Profile specifies Central recorder Recording Enabled Gateway Gateway Preferred SIP Trunk ICT Recording ISR-G2/ CUBE 5. V Remote Device Profile User 2 1. IP phone User 1 User central recorder branch recorder [1] [5]: An call from User 1 on IP phone (on Cluster 2) to User 2, is extended off-network via User 2 Remote Device Profile to a remote destination. The call is routed through Cluster 1 to egress to the PSTN via the SIP ICT trunk. The call is answered by User 2 on the mobile remote destination phone. [6]: Recording is started for User 2 Remote Device Profile on Cluster 2. Cluster 2 requests Cluster 1 to start a gateway recording session. [7]: Cluster 1 starts the recording session and instructs the gateway to fork media to the central recorder. 146

147 Inter-cluster Recording IP Phone call from cluster 2 to Remote Device Profile on Cluster 1 Two recording sessions with both Gateway and Phone selected Recording Enabled Gateway ISR-G2/ CUBE 5. V Gateway Preferred Recording 3. Remote Device Profile User 2 SIP Trunk ICT Phone Preferred Recording IP phone User User 2 central recorder branch recorder [1] [5]: An call from a IP phone User 1 (in Cluster 2) to User 2 (in Cluster 1) is extended off-network via User 2 Remote Device Profile to a remote destination. The call is routed through Cluster 1 to egress to the PSTN via the SIP ICT trunk. The call is answered by User 2 on the mobile remote destination phone. [6]: Recording is started for User 1. Phone source is Gateway Preferred, but the call is connected to User 2 (RDP). From the Phone perspective, the gateway is not in the call flow so the Phone is selected to fork the media to the branch recorder. [7]: Recording is started for User 2. Cluster 1 starts the recording session and instructs the gateway to fork media to the central recorder. 147

148 Inter-cluster Recording External call from Cluster 1 to mobile remote destination on Cluster 2 ingress gateway selected, the recording profile specifies central recorder Recording Enabled Gateway SIP Trunk ICT Gateway Preferred Recording ISR-G2/ CUBE V V GW central recorder branch recorder [1] [5]: An external call from PSTN via the gateway in Cluster 1 to a mobile remote destination profile in Cluster 2 extended to remote destination mobile phone through a gateway in Cluster 2, mobile answers the call [6]: Recording is started from the remote destination profile in Cluster 2. Cluster 2 request Cluster 1 to start a gateway recording session [7] Cluster 1 starts the recording session and instructs the gateway in Cluster 1 to fork the media to central recorder. 148

149 Recording Call Flows Multi-Cluster IP Phone Including Mid-Call features: Hold/Resume, Shared-lines Transfer Conference Automatic recording mode is used for illustrations. Selective recording modes are supported unless otherwise noted.

150 Inter-cluster Recording External call from Cluster 1 is HELD by User on Cluster 2 Mid-Call - Hold Recording Enabled Gateway V ISR- G2/CUBE Cluster 1 Cluster SIP Trunk ICT MOH 2. Gateway Preferred Recording 1.user holds the call User 1 central recorder branch recorder [1] [5]: User 1 puts the call on hold; the media stream is stopped [6]: Unified CM in Cluster 2 plays MOH to the caller [7] [8]: Unified CM in Cluster 2 requests Cluster 1 to stop the recording session. Cluster 1 instructs the gateway to stop the session. 150

151 Inter-cluster Recording External call from Cluster 1 is RESUMED by User on Cluster 2 Mid-call - Resume from same device Recording Enabled Gateway V ISR- G2/CUBE Cluster 1 Cluster SIP Trunk ICT MOH 11. Gateway Preferred Recording User user resumes the call central recorder branch recorder [9] [14]: User 1 on Cluster 2 resumes the call; Cluster 2 stops playing MOH and reconnects the External Caller. [15]: Cluster 2 requests Cluster 1 to (re)start the recording. [16]: Cluster 1 instructs the gateway for fork media. The Recording Profile for User 1 IP Phone is configured to use the central recorder. 151

152 Inter-cluster Recording External call from Cluster 1 is RESUMED by User on Cluster 2 Mid-call - Resume from shared line on different device with gateway selected Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Gateway Preferred Recording V ISR- G2/CUBE MOH 11. User 1 Gateway Preferred Recording User 2 central recorder branch recorder 9.user 2 resumes the call from shared line [9] [14]: User 2 on Cluster 2 resumes the call from a shared line; Cluster 2 stops playing MOH and reconnects User 2 with the external caller [15]: Cluster 2 requests Cluster 1 to (re)start the recording [16]: Cluster 1 instructs the gateway to (re)establish the recording session with the branch recorder. The Recording Profile for User 2 IP Phone is configured to use the branch recorder. 152

153 Inter-cluster Recording External call from Cluster 1 is RESUMED by User on Cluster 2 Mid-call - Resume from shared line on different device with phone selected Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Gateway Preferred Recording ISR- G2/CUBE V MOH 11. User 1 Phone Preferred Recording 14. User 2 central recorder branch recorder user 2 resumes the call from shared line [9] [14]: User 2 on Cluster 2 resumes the call from a shared line; Cluster 2 stops playing MOH and reconnects the User with the external caller [15]: User 2 IP Phone is configured Phone preferred, so the Phone is selected to fork media to the branch recorder. The Recording Profile for User 2 IP Phone is configured to use the branch recorder. 153

154 Inter-cluster Recording External call from Cluster 1 to IP Phone on Cluster 2 Mid-Call - Transfer/Consult Recording Enabled Gateway ISR- G2/CUBE V Cluster 1 Cluster SIP Trunk ICT MOH Gateway Preferred Recording 12. User 1 1.presses xfer softkey 9.dials User User 2 central recorder branch recorder [1] [6]: User 1 presses the transfer softkey; the call is put on hold; the media stream is stopped; Cluster 2 plays MOH to the External Caller [7] [8]: Cluster 2 requests Cluster 1 to stop the recording; Cluster 1 instructs the gateway to stop the session. [9] [14]: User 1 starts a consultation call with User 2; both IP phones start new recording sessions for the consult call. The Phone is selected to fork the media to the branch recorder (as configured in the recording profile). 154

155 Inter-cluster Recording External call from Cluster 1 to IP Phone on Cluster 2 Mid-Call - Transfer/Consult Recording Enabled Gateway ISR- G2/CUBE V Cluster 1 Cluster SIP Trunk ICT MOH Gateway Preferred Recording 19. User 1 15.presses xfer softkey 21. User 2 central recorder branch recorder [15] [21]: User 1 on Cluster 2 presses the softkey again to complete the transfer; the Consult calls ends, both Phone recording sessions stop and MOH stops playing to the external caller. [22] [25]: The External Caller is connected to User 2 [26]: User 2 Phone is set to Gateway preferred, so Cluster 2 requests Cluster 1 to start the recording session [27]: Cluster 1 instructs the gateway to fork media to the central recorder 155

156 Inter-cluster Recording Internal call with consult to external party Mid-Call - Transfer/Consult Recording Enabled Gateway Cluster 1 Cluster 2 SIP Trunk ICT Gateway Preferred Recording ISR- G2/CUBE 11 V MOH User 2 External Party 7. central recorder User 1 Gateway Preferred Recording 1.presses xfer softkey 8.dials external number branch recorder [1] [6]: User 1 presses the transfer softkey which puts User 2 on hold; MOH is played from Cluster 1 to User 2. [7]: The recording session for the internal call is stopped. The recording session for User 2 in Cluster 2 continues [8] [11]: User 1 makes a consultation call to an external party [12]: A new recording session is started for User 1; the Gateway is selected to fork the media. 156

157 Inter-cluster Recording Internal call with consult to external party Mid-Call - Transfer/Consult Recording Enabled Gateway ISR- G2/CUBE V Cluster 1 Cluster MOH SIP Trunk ICT Gateway Preferred Recording User External Party central recorder User 1 13 User press the transfer softkey again branch recorder [13] [18]: User 1 presses the softkey again to complete the transfer. Cluster 1 stops playing MOH, ends the consultation call, and the recording session for the consultation call is stopped. [19] [21]: Cluster 1 joins User 2 with the External Party. [22] [23]: The recording session for User 2 is restarted, The recording session for the consult call with User 1 is stopped. Cluster 2 requests Cluster 1 to start a new recording session. [24]: Cluster 1 instructs gateway to fork media to the central recorder 157

158 Additional Secure Recording Call Flows

Recording. Recording Overview

Recording. Recording Overview Overview, page 1 Prerequisites, page 3 Configuration Task Flow, page 3 Call Flow Examples, page 16 Interactions and Restrictions, page 16 Overview Call recording is a Cisco Unified Communications Manager

More information

Following configurations are needed regardless of the recording technology being used.

Following configurations are needed regardless of the recording technology being used. PBX Configuration CuCM configuration SIP Trunk Following configurations are needed regardless of the recording technology being used. 1. Create a new SIP Trunk Security Profile named "Imagicle Call Recording

More information

Internet Protocol Version 6 (IPv6)

Internet Protocol Version 6 (IPv6) This chapter provides information about Internet Protocol version 6 (IPv6), which is the latest version of the Internet Protocol (IP). Packets are used to exchange data, voice, and video traffic over dual-stack

More information

Monitoring and Recording

Monitoring and Recording CHAPTER 34 Call centers need to be able to guarantee the quality of customer service that an agent in a call center provides. To protect themselves from legal liability, call centers need to be able to

More information

Services Extended Media Forking

Services Extended Media Forking Cisco Unified Communications Gateway Services--Extended Media Forking The Cisco Unified Communications (UC) Services API provides a unified web service interface for the different services in IOS gateway

More information

Cisco Unified Communications Gateway Services--Extended Media Forking

Cisco Unified Communications Gateway Services--Extended Media Forking Cisco Unified Communications Gateway Services--Extended Media Forking The Cisco Unified Communications (UC) Services API provides a unified web service interface for the different services in IOS gateway

More information

Internet Protocol Version 6 (IPv6)

Internet Protocol Version 6 (IPv6) CHAPTER 29 Internet Protocol version 6 (IPv6), which is the latest version of the Internet Protocol (IP) that uses packets to exchange data, voice, and video traffic over digital networks, increases the

More information

System-Level Configuration Settings

System-Level Configuration Settings CHAPTER 5 Configure system-level settings before you add devices and configure other Cisco Unified CallManager features. This section covers the following topics: Server Configuration, page 5-1 Cisco Unified

More information

Music On Hold. Configuration Checklist for Music On Hold CHAPTER

Music On Hold. Configuration Checklist for Music On Hold CHAPTER CHAPTER 36 The integrated (MOH) feature allows users to place on-net and off-net users on hold with music that is streamed from a streaming source. The feature allows two types of hold: End-user hold Network

More information

Cisco Unified Communication IOS Services API

Cisco Unified Communication IOS Services API CHAPTER 1 This chapter describes the Cisco Unified Communication IOS Services Application Programming Interface (CUCISAPI). The CUCISAPI enables the development of advanced Cisco Unified Communication

More information

Calabrio Recording Services. Deployment Guide for Cisco MediaSense

Calabrio Recording Services. Deployment Guide for Cisco MediaSense Calabrio Recording Services Deployment Guide for Cisco MediaSense Version First Published: September 30, 2014 Last Updated: June 3, 2014 Calabrio and Calabrio ONE are registered trademarks and the Calabrio

More information

Customer Guide to Cisco MediaSense Integrations. March

Customer Guide to Cisco MediaSense Integrations. March Customer Guide to Cisco MediaSense Integrations March 2017 www.incontact.com Introduction Customer Guide to Cisco MediaSense Integrations Version: This guide should be used with NICE Uptivity (formerly

More information

BT SIP Trunk Configuration Guide

BT SIP Trunk Configuration Guide CUCM 9.1 BT SIP Trunk Configuration Guide This document covers service specific configuration required for interoperability with the BT SIP Trunk service. Anything which could be considered as normal CUCM

More information

On-Site 911 Notification Using Cisco Unified Communications BRKUCC-2012

On-Site 911 Notification Using Cisco Unified Communications BRKUCC-2012 On-Site 911 Notification Using Cisco Unified Communications Session Objective This session will illustrate different methods by which an enterprise can enhance the emergency call handling At the end of

More information

Customer Guide to Cisco JTAPI- BiB Integrations. March

Customer Guide to Cisco JTAPI- BiB Integrations. March Customer Guide to Cisco JTAPI- BiB Integrations March 2017 www.incontact.com Introduction Customer Guide to Cisco JTAPI-BiB Integrations Version: This guide should be used with NICE Uptivity (formerly

More information

Implementing Cisco Unified Communications Manager Part 2, Volume 1

Implementing Cisco Unified Communications Manager Part 2, Volume 1 Implementing Cisco Unified Communications Manager Part 2, Volume 1 Course Introduction Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your Training Curriculum

More information

Client services framework setup

Client services framework setup In Cisco Unified Communications Manager Administration, use the Device > Phone menu path to configure the Cisco Unified Client Services Framework device. This section describes how to configure a Cisco

More information

Mobile Agent. Capabilities. Cisco Unified Mobile Agent Description. Unified Mobile Agent Provides Agent Sign-In Flexibility

Mobile Agent. Capabilities. Cisco Unified Mobile Agent Description. Unified Mobile Agent Provides Agent Sign-In Flexibility Capabilities, page 1 Initial setup, page 7 Administration and usage, page 17 Capabilities Cisco Unified Description Unified supports call center agents using phones that Packaged CCE does not directly

More information

Cisco Unified Communications Manager 9.0

Cisco Unified Communications Manager 9.0 Data Sheet Cisco Unified Communications Manager 9.0 Cisco Unified Communications Manager is the heart of Cisco collaboration services, enabling session and call control for video, voice, messaging, mobility,

More information

Cisco Unified Mobility

Cisco Unified Mobility CHAPTER 14 extends the rich call control capabilities of Cisco Unified Communications Manager from the primary workplace desk phone of a mobile worker to any location or device of their choosing. For example,

More information

Vendor: Cisco. Exam Code: Exam Name: Troubleshooting Cisco IP Telephony and Video (CTCOLLAB) Version: Demo

Vendor: Cisco. Exam Code: Exam Name: Troubleshooting Cisco IP Telephony and Video (CTCOLLAB) Version: Demo Vendor: Cisco Exam Code: 300-080 Exam Name: Troubleshooting Cisco IP Telephony and Video (CTCOLLAB) Version: Demo DEMO QUESTION 1 Which four performance counters are available when monitoring a Cisco MTP

More information

map q850-cause through mgcp packagecapability

map q850-cause through mgcp packagecapability map q850-cause through mgcp package-capability map q850-cause through mgcp packagecapability 1 map q850-cause map q850-cause through mgcp package-capability map q850-cause To play a customized tone to

More information

Silent Monitoring. Silent Monitoring Overview

Silent Monitoring. Silent Monitoring Overview Overview, page 1 Prerequisites, page 2 Configure Task Flow, page 2 Interactions and Restrictions, page 8 Overview Silent call monitoring allows a supervisor to eavesdrop on a phone conversation. The most

More information

BRKCOC-2399 Inside Cisco IT: Integrating Spark with existing large deployments

BRKCOC-2399 Inside Cisco IT: Integrating Spark with existing large deployments Inside Cisco IT: Integrating Spark with existing large deployments Jan Seynaeve, Sr. Collaborations Engineer Luke Clifford, Sr. Collaborations Engineer Cisco Spark How Questions? Use Cisco Spark to communicate

More information

Application Notes for Configuring Tidal Communications tnet Business VoIP with Avaya IP Office using SIP Registration - Issue 1.0

Application Notes for Configuring Tidal Communications tnet Business VoIP with Avaya IP Office using SIP Registration - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Tidal Communications tnet Business VoIP with Avaya IP Office using SIP Registration - Issue 1.0 Abstract These Application Notes

More information

Cisco Unified MeetingPlace Integration

Cisco Unified MeetingPlace Integration CHAPTER 14 This chapter covers system-level design and implementation of Cisco Unified MeetingPlace 5.4 in a Cisco Unified Communications Manager 5.x environment. The following aspects of design and configuration

More information

Infrastructure Configuration Product Fields

Infrastructure Configuration Product Fields Infrastructure Configuration Product s Infrastructure Data Object s, page 1 Infrastructure Data Object s To create Configuration Templates, you must add infrastructure Configuration Products to the Configuration

More information

Conference Now. Conference Now Overview. Conference Now Prerequisites

Conference Now. Conference Now Overview. Conference Now Prerequisites Overview, page 1 Prerequisites, page 1 Task Flow, page 2 Interactions and Restrictions, page 6 Overview The feature allows both external and internal callers to join a conference by dialing a IVR Directory

More information

Command or Action Step 1. Create and Configure Cisco Jabber Devices, on page 1. Configure a SIP Trunk, on page 6

Command or Action Step 1. Create and Configure Cisco Jabber Devices, on page 1. Configure a SIP Trunk, on page 6 s Workflow, page 1 s Workflow Command or Action Purpose Create and Configure Cisco Jabber Devices, on page 1 Create at least one device for every user that will access Cisco Jabber. Configure a SIP Trunk,

More information

Cisco Unified CME Commands: M

Cisco Unified CME Commands: M Cisco Unified CME Commands: M mac-address (ephone), page 3 mac-address (voice-gateway), page 5 mailbox-selection (dial-peer), page 7 mailbox-selection (ephone-dn), page 9 max-calls-per-button, page 11

More information

Computer Telephony Integration

Computer Telephony Integration This chapter provides information about Computer telephony integration (CTI) which enables you to leverage computer-processing functions while making, receiving, and managing telephone calls. CTI applications

More information

Universal device template setup

Universal device template setup This chapter contains information to set up universal device templates. About universal device template display preference setup, page 1, page 2 About universal device template display preference setup

More information

PASS4TEST. IT Certification Guaranteed, The Easy Way! We offer free update service for one year

PASS4TEST. IT Certification Guaranteed, The Easy Way!   We offer free update service for one year PASS4TEST IT Certification Guaranteed, The Easy Way! \ http://www.pass4test.com We offer free update service for one year Exam : 642-444 Title : IP Telephony Exam (CIPT) Vendors : Cisco Version : DEMO

More information

CUCM 10.5 / CUBE 9.5. BT SIP Trunk Configuration Guide. 1 BT SIP Trunk Configuration Guide

CUCM 10.5 / CUBE 9.5. BT SIP Trunk Configuration Guide. 1 BT SIP Trunk Configuration Guide 1 BT SIP Trunk Configuration Guide CUCM 10.5 / CUBE 9.5 BT SIP Trunk Configuration Guide This document covers service specific configuration required for interoperability with the BT SIP Trunk service.

More information

Product Datasheet telisca Recording

Product Datasheet telisca Recording Product Datasheet telisca Directory Phone Directory Jabber UDS Server Web Directory IPS Popup / Reverse Lookup Personal Directory H350 Video Conf directory Corporate Speed Dials ClickNDial Alerting Voice

More information

Configure Cisco IP Phones

Configure Cisco IP Phones Cisco IP Phones Overview, page 1 Cisco IP Phones Configuration Task Flow, page 1 Cisco IP Phones Overview Cisco Unified IP Phones are full-featured telephones that provide voice communication over an IP

More information

Phone template. Add phones to database. Procedure

Phone template. Add phones to database. Procedure Cisco Unified Communications Manager Bulk Administration (BAT) gives the administrator a fast and efficient way to add, update, or delete large numbers of phones in batches, rather than performing individual

More information

CAPPS: Implementing Cisco Collaboration Applications v1

CAPPS: Implementing Cisco Collaboration Applications v1 Course Objectives Implement Cisco Unity Connection in a Cisco Unified Communications Manager deployment Describe how to implement Cisco Unity Express in a Cisco Unified Communications Manager Express deployment

More information

Cisco Unified IP Phone setup

Cisco Unified IP Phone setup This chapter provides information about working with and configuring Cisco Unified IP Phones in Cisco Unified Communications Manager Administration. About Cisco Unified IP Phones and device setup, page

More information

This chapter provides information about using Cisco Unified Communications Manager for working with and configuring Cisco gateways.

This chapter provides information about using Cisco Unified Communications Manager for working with and configuring Cisco gateways. This chapter provides information about using Cisco Unified Communications Manager for working with and configuring Cisco gateways. About gateway setup, page 1 Gateway reset, page 2 Gateway deletion, page

More information

Cisco.Certkiller v by.Luger.57q. Exam Code:

Cisco.Certkiller v by.Luger.57q. Exam Code: Cisco.Certkiller.642-447.v2013-12-24.by.Luger.57q Number: 642-447 Passing Score: 825 Time Limit: 120 min File Version: 16.5 http://www.gratisexam.com/ Exam Code: 642-447 Exam Name: Cisco CIPT1 v8.0 Implementing

More information

Configure Voice and Video Communication

Configure Voice and Video Communication s for On-Premises Deployments, page 1 for Cloud-Based Deployments, page 23 s for On-Premises Deployments Command or Action Purpose Install Cisco Options Package File for Devices, on page 2. Complete this

More information

JPexam. 最新の IT 認定試験資料のプロバイダ IT 認証であなたのキャリアを進めます

JPexam.   最新の IT 認定試験資料のプロバイダ IT 認証であなたのキャリアを進めます JPexam 最新の IT 認定試験資料のプロバイダ http://www.jpexam.com IT 認証であなたのキャリアを進めます Exam : 642-427 Title : Troubleshooting Cisco Unified Communications v8.0 (TVOICE v8.0) Vendor : Cisco Version : DEMO Get Latest & Valid

More information

Exam Questions Demo Cisco. Exam Questions CCIE Collaboration.

Exam Questions Demo   Cisco. Exam Questions CCIE Collaboration. Cisco Exam Questions 400-051 CCIE Collaboration Version:Demo 1. In Cisco IOS routers that use low latency queuing, which algorithm is used to presort traffic going into the default queue? A. first-in,

More information

Cisco Unified IP Phone setup

Cisco Unified IP Phone setup Cisco Unified IP Phone setup This chapter provides information about working with and configuring Cisco Unified IP Phones in Cisco Unified Communications Manager Administration. About Cisco Unified IP

More information

Setting Up a Mitel SX-2000 Digital PIMG Integration with Cisco Unity Connection

Setting Up a Mitel SX-2000 Digital PIMG Integration with Cisco Unity Connection Up a Mitel SX-2000 Digital PIMG Integration with Cisco Unity Connection Up a Mitel SX-2000 Digital PIMG Integration, page 1 Up a Mitel SX-2000 Digital PIMG Integration Task List for Mitel SX-2000 PIMG

More information

Cisco Contact Center Express 10.0: Feature Design, Deployment, and Troubleshooting

Cisco Contact Center Express 10.0: Feature Design, Deployment, and Troubleshooting Cisco Contact Center Express 10.0: Feature Design, Deployment, and Troubleshooting Ron Rodriguez, Technical Solutions Manager CBABU Mike Turnbow, Technical Solutions Manager CBABU Agenda CCX 10.0 Feature

More information

PracticeTorrent. Latest study torrent with verified answers will facilitate your actual test

PracticeTorrent.   Latest study torrent with verified answers will facilitate your actual test PracticeTorrent http://www.practicetorrent.com Latest study torrent with verified answers will facilitate your actual test Exam : 300-070 Title : Implementing Cisco IP Telephony & Video, Part 1 v1.0 Vendor

More information

Cisco Unified IP Phone Configuration

Cisco Unified IP Phone Configuration CHAPTER 67 Cisco Unified IP Phones as full-featured telephones can plug directly into your IP network. You use the Cisco Unified Communications Manager Administration Phone Configuration window to configure

More information

Cisco Mobility. Cisco Unified Mobility. Configure Cisco Unified Mobility. Cisco Unified Mobility, page 1 Cisco Jabber for Mobile, page 66

Cisco Mobility. Cisco Unified Mobility. Configure Cisco Unified Mobility. Cisco Unified Mobility, page 1 Cisco Jabber for Mobile, page 66 Cisco Unified Mobility, page 1 Cisco Jabber for Mobile, page 66 Cisco Unified Mobility This chapter provides information about Cisco Unified Mobility which extends the rich call control capabilities of

More information

Application Notes for Phonect SIP Trunk Service and Avaya IP Office 7.0 Issue 1.0

Application Notes for Phonect SIP Trunk Service and Avaya IP Office 7.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Phonect SIP Trunk Service and Avaya IP Office 7.0 Issue 1.0 Abstract These Application Notes describe the procedures for configuring Session

More information

Cisco Unified CM SIP Trunking, Session Management, and Global Dial Plan Replication

Cisco Unified CM SIP Trunking, Session Management, and Global Dial Plan Replication LTRUCC-2150 Cisco Unified CM SIP Trunking, Session Management, and Global Dial Plan Replication Paul Giralt - @PaulGiralt Markus Schneider - @Markus73 Agenda Objectives Technology Overview Unified CM Session

More information

Unified Contact Center Express Release Notes 10.6(1)

Unified Contact Center Express Release Notes 10.6(1) Unified Contact Center Express Release Notes 10.6(1) Introduction, page 1 New and Updated Features, page 2 Limitations and Restrictions, page 3 Caveats, page 8 Documentation Feedback, page 9 Documentation

More information

Extend and Connect. Extend and Connect Overview

Extend and Connect. Extend and Connect Overview Overview, page 1 Prerequisites, page 2 Configuration Task Flow, page 2 Interactions and Restrictions, page 8 Overview The feature allows administrators to deploy Unified Communications (UC) Computer Telephony

More information

Unified Border Element (CUBE) with Cisco Unified Communications Manager (CUCM) Configuration Example

Unified Border Element (CUBE) with Cisco Unified Communications Manager (CUCM) Configuration Example Unified Border Element (CUBE) with Cisco Unified Communications Manager (CUCM) Configuration Example Document ID: 99863 Contents Introduction Prerequisites Requirements Components Used Conventions Configure

More information

CISCO EXAM QUESTIONS & ANSWERS

CISCO EXAM QUESTIONS & ANSWERS CertifyMe Number: 642-427 Passing Score: 800 Time Limit: 120 min File Version: 29.0 http://www.gratisexam.com/ CISCO 642-427 EXAM QUESTIONS & ANSWERS Exam Name: TVOICE v8.0 Troubleshooting Cisco Unified

More information

Hold Reversion. Configuration Checklist for Hold Reversion CHAPTER

Hold Reversion. Configuration Checklist for Hold Reversion CHAPTER CHAPTER 25 The hold reversion feature alerts a phone user when a held call exceeds a configured time limit. This chapter provides information on the following topics: Configuration Checklist for, page

More information

Deploying Harmony Workforce Optimization on Unified Contact Center Enterprise

Deploying Harmony Workforce Optimization on Unified Contact Center Enterprise Deploying Harmony Workforce Optimization on Unified Contact Center Enterprise Release 5.2.2 Issue 1 September 2018 Contents Chapter 1: Introduction... 8 Purpose... 8 Prerequisites... 8 Chapter 2: Harmony

More information

Partitioned Intradomain Federation for IM and Presence Service on Cisco Unified Communications Manager, Release 11.5(1)SU2

Partitioned Intradomain Federation for IM and Presence Service on Cisco Unified Communications Manager, Release 11.5(1)SU2 Partitioned Intradomain Federation for IM and Presence Service on Cisco Unified Communications Manager, First Published: 2017-01-10 Americas Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose,

More information

Call Recording. Imagicle. Never miss a word. Imagicle. ApplicationSuite INCLUDED INTO THE FOR CISCO UC

Call Recording. Imagicle. Never miss a word. Imagicle. ApplicationSuite INCLUDED INTO THE FOR CISCO UC ApplicationSuite FOR CISCO UC Never miss a word. Copyright spa 2010-2018. Brands cited must and will be considered as registered brands property of their respective owners. Record and archive your calls

More information

Cisco Unified Communications Manager Trunks

Cisco Unified Communications Manager Trunks CHAPTER 2 A trunk is a communications channel on Cisco Unified Communications Manager (Cisco Unified CM) that enables Cisco Unified CM to connect to other servers. Using one or more trunks, Cisco Unified

More information

Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise

Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise The Changing Landscape IP-based unified communications is widely deployed in enterprise networks, both for internal calling

More information

Cisco UCCX Configuration Guide. Comstice Mobile Agent App for Cisco UCCX Configuration Steps. made with

Cisco UCCX Configuration Guide. Comstice Mobile Agent App for Cisco UCCX Configuration Steps. made with Cisco UCCX Configuration Guide Comstice Mobile Agent App for Cisco UCCX Configuration Steps made with Mobile Agent App Benefits Contact Center Agent without Desktop PC Comstice Mobile Agent App is a Cisco

More information

Setting up Alcatel 4400 Digital PIMG Integration

Setting up Alcatel 4400 Digital PIMG Integration up Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection, on page 1 Up an Alcatel 4400 Digital PIMG Integration with

More information

Setting Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection

Setting Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection up Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection, page 1 Up an Alcatel 4400 Digital PIMG Integration with Cisco

More information

Designing and deploying UC networks with Cisco Unified Session Management Edition

Designing and deploying UC networks with Cisco Unified Session Management Edition Designing and deploying UC networks with Cisco Unified Session Management Edition Tony Mulchrone Technical Marketing Engineer Cisco Collaboration Technology Group Housekeeping We value your feedback- don't

More information

Course Outline: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1)

Course Outline: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) Course Outline: Implementing Cisco IP Telephony & Video, Part 1 (CIPTV1) Learning Method: Instructor-led Classroom Learning Duration: 5.00 Day(s)/ 40 hrs : CIPTV1 v1.0 gives the learner all the tools they

More information

AT&T IP Flexible Reach And IP Toll Free Cisco Unified Communication Manager H.323 Configuration Guide. Issue /3/2008

AT&T IP Flexible Reach And IP Toll Free Cisco Unified Communication Manager H.323 Configuration Guide. Issue /3/2008 AT&T IP Flexible Reach And IP Toll Free Cisco Unified Communication Manager H.323 Configuration Guide Issue 2.17 3/3/2008 Page 1 of 49 TABLE OF CONTENTS 1 Introduction... 4 2 Special Notes... 4 3 Overview...

More information

Deployment Models. Cisco Unified Contact Center Enterprise Solution Reference Network Design, Release 9.x 1

Deployment Models. Cisco Unified Contact Center Enterprise Solution Reference Network Design, Release 9.x 1 There are numerous ways that Unified Contact Center Enterprise (Unified CCE) can be deployed, but the deployments can generally be categorized into the following major types or models: Single Site Multisite

More information

IP Addressing Modes for Cisco Collaboration Products

IP Addressing Modes for Cisco Collaboration Products IP Addressing Modes for Cisco Collaboration Products IP Addressing Modes, page 1 Recommended IPv6 Addressing Modes for CSR 12.0 Products, page 3 IPv6 Addressing in Cisco Collaboration Products, page 9

More information

Q&As. Implementing Cisco Collaboration Devices v1.0. Pass Cisco Exam with 100% Guarantee

Q&As. Implementing Cisco Collaboration Devices v1.0. Pass Cisco Exam with 100% Guarantee 210-060 Q&As Implementing Cisco Collaboration Devices v1.0 Pass Cisco 210-060 Exam with 100% Guarantee Free Download Real Questions & Answers PDF and VCE file from: 100% Passing Guarantee 100% Money Back

More information

Bandwidth, Latency, and QoS for Core Components

Bandwidth, Latency, and QoS for Core Components Bandwidth, Latency, and QoS for Core Components, on page 1 Bandwidth, Latency, and QoS for Optional Cisco Components, on page 18 Bandwidth, Latency, and QoS for Optional Third-Party Components, on page

More information

Leveraging Amazon Chime Voice Connector for SIP Trunking. March 2019

Leveraging Amazon Chime Voice Connector for SIP Trunking. March 2019 Leveraging Amazon Chime Voice Connector for SIP Trunking March 2019 Notices Customers are responsible for making their own independent assessment of the information in this document. This document: (a)

More information

ID Features Tested Case Title Description Call Component Flow Status Defects UC802CL.ACE.001 Basic Call Flow Integrate Cisco Application Control

ID Features Tested Case Title Description Call Component Flow Status Defects UC802CL.ACE.001 Basic Call Flow Integrate Cisco Application Control Application Control Engine System Test Results for Contact Center, Cisco Unified System Release 8.0(2) UC802CL.ACE.001 Basic Call Flow Integrate Cisco Application Control Verifies that intelligent loadbalancing

More information

IP Addressing Modes for Cisco Collaboration Products

IP Addressing Modes for Cisco Collaboration Products IP Addressing Modes for Cisco Collaboration Products IP Addressing Modes, on page 1 Recommended IPv6 Addressing Modes for CSR 12.1/12.0 Products, on page 2 IPv6 Addressing in Cisco Collaboration Products,

More information

Cisco Unified Mobility Advantage and Cisco Unified Mobile Communicator Integration

Cisco Unified Mobility Advantage and Cisco Unified Mobile Communicator Integration CHAPTER 15 Cisco Unified Mobility Advantage and Cisco Unified Mobile Communicator Integration Cisco Unified Communications Manager provides certain functionality for Cisco Unified Mobile Communicator clients

More information

Release Notes for Cisco Finesse Release 9.0(1)

Release Notes for Cisco Finesse Release 9.0(1) These release notes provide the following information. You might need to notify your users about some of the information provided in this document. Introduction, page 1 Hardware and Software Specifications

More information

Mobile Agent. Capabilities. Cisco Unified Mobile Agent Description. Capabilities, page 1 Initial Setup, page 17 Administration and Usage, page 29

Mobile Agent. Capabilities. Cisco Unified Mobile Agent Description. Capabilities, page 1 Initial Setup, page 17 Administration and Usage, page 29 Capabilities, page 1 Initial Setup, page 17 Administration and Usage, page 29 Capabilities Cisco Unified Description enables an agent to use any PSTN phone and a broadband VPN connection (for agent desktop

More information

Innovation Networking App Note

Innovation Networking App Note Innovation Networking App Note G12 Communications ShoreTel and G12 Communications for SIP Trunking (Native) 1 (877) 311-8750 sales@g12com.com Jackson St. #19390, Seattle, WA 98104 Product: ShoreTel G12

More information

Jabber for Windows - Quick Start Guide

Jabber for Windows - Quick Start Guide Jabber for Windows - Quick Start Guide Contents Introduction Prerequisites Software Requirements Hardware Requirements Configuring Phone Services Jabber Softphone Jabber Deskphone Deskphone Configuration

More information

Video Telephony. Configure Video Telephony

Video Telephony. Configure Video Telephony This chapter provides information about video telephony. Cisco Unified Communications Manager supports video telephony and thus unifies the world of voice and video calls. Video endpoints use Cisco Unified

More information

Gateway Options. PSTN Gateway, page 2

Gateway Options. PSTN Gateway, page 2 Cisco offers a large range of voice gateway models to cover a large range of requirements. Many, but not all, of these gateways have been qualified for use with Unified CVP. For the list of currently supported

More information

Configuration Limits and Feature Availability for Reference Designs

Configuration Limits and Feature Availability for Reference Designs Configuration Limits and Feature Availability for s Configuration Limits, page 1 Feature Availability for s, page 13 Configuration Limits The following tables list key configuration limits for Contact

More information

Cisco Exam Questions & Answers

Cisco Exam Questions & Answers Cisco 642-457 Exam Questions & Answers Number: 642-457 Passing Score: 800 Time Limit: 120 min File Version: 35.5 http://www.gratisexam.com/ Sections 1. 1-18 2. 19-36 3. 37-54 4. 55-72 Cisco 642-457 Exam

More information

Cisco Unified Survivable Remote Site Telephony and Cisco Unified Enhanced Survivable Remote Site Telephony Version 11.0

Cisco Unified Survivable Remote Site Telephony and Cisco Unified Enhanced Survivable Remote Site Telephony Version 11.0 Data Sheet Cisco Unified Survivable Remote Site Telephony and Cisco Unified Enhanced Survivable Remote Site Telephony Version 11.0 Helping Provide Reliable Communications to Branch Offices, Teleworkers,

More information

An Overview of Cisco MobilityManager

An Overview of Cisco MobilityManager CHAPTER 1 An Overview of Cisco MobilityManager This chapter describes Cisco MobilityManager and includes these sections: Key Features and Benefits, page 1-4 Use Case Examples, page 1-5 Compatibility with

More information

Design Considerations for Integrated Features

Design Considerations for Integrated Features Agent Greeting Considerations, page 1 Cisco Outbound Option Considerations, page 5 Courtesy Callback Considerations, page 22 Call Context Considerations, page 28 Mixed Codec Considerations, page 29 Mobile

More information

Cisco TelePresence Conductor with Unified CM

Cisco TelePresence Conductor with Unified CM Cisco TelePresence Conductor with Unified CM Deployment Guide TelePresence Conductor XC3.0 Unified CM 10.x Revised February 2015 Contents Introduction 5 About this document 5 Related documentation 5 About

More information

INTEROPERABILITY REPORT

INTEROPERABILITY REPORT [ ] INTEROPERABILITY REPORT Ascom IP-DECT Cisco Unified Communcations Manager, version 8.6.1.20000-1 IP PBX Integration Session Initiation Protocol (SIP) Ascom IP-DECT R5.1 Ascom January 2013 Interoperability

More information

Unified Communication Platform

Unified Communication Platform fonouc Unified Communication Platform fonouc Unified Communications Service Platform, is a scalable, managed, turnkey solution for carries and service providers, designed to provide multi-tenant business

More information

INTEROPERABILITY REPORT

INTEROPERABILITY REPORT [ ] INTEROPERABILITY REPORT Ascom IP-DECT Cisco Unified Communcations Manager, version 8.6.1.20000-1 IP PBX Integration Session Initiation Protocol (SIP) Ascom, Gothenburg, SE March, 2012 Interoperability

More information

Call Control Discovery

Call Control Discovery CHAPTER 3 The call control discovery feature leverages the Service Advertisement Framework (SAF) network service, a proprietary Cisco service, to facilitate dynamic provisioning of inter-call agent information.

More information

Application Notes for Configuring CenturyLink SIP Trunking with Avaya IP Office Issue 1.0

Application Notes for Configuring CenturyLink SIP Trunking with Avaya IP Office Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring CenturyLink SIP Trunking with Avaya IP Office 6.1 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Product Overview. Benefits CHAPTER

Product Overview. Benefits CHAPTER CHAPTER 1 Revised July 3, 2012 The Cisco TelePresence Exchange System is an integrated video service-creation platform that enables service providers and strategic partners to offer secure cloud-based

More information

Application Notes for Configuring Windstream SIP Trunking with Avaya IP Office - Issue 1.0

Application Notes for Configuring Windstream SIP Trunking with Avaya IP Office - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Windstream SIP Trunking with Avaya IP Office - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Cisco TelePresence Conductor with Cisco Unified Communications Manager

Cisco TelePresence Conductor with Cisco Unified Communications Manager Cisco TelePresence Conductor with Cisco Unified Communications Manager Deployment Guide TelePresence Conductor XC4.0 Unified CM 10.5(2) January 2016 Contents Introduction 6 About this document 6 Related

More information

map q850-cause through mgcp package-capability

map q850-cause through mgcp package-capability map q850-cause through mgcp package-capability map q850-cause, page 4 map resp-code, page 6 max1 lookup, page 9 max1 retries, page 11 max2 lookup, page 13 max2 retries, page 15 max-bandwidth, page 17 max-calls,

More information

Configure Conference Bridges

Configure Conference Bridges Conference Bridges Overview, page 1 Conference Bridge Types, page 1 Call Preservation, page 4 Call Preservation Scenarios, page 5 Conference Bridge Configuration Task Flow, page 7 Conference Bridges Overview

More information

INTEGRATING CISCO UNIFIED COMMUNICATIONS APPLICATIONS

INTEGRATING CISCO UNIFIED COMMUNICATIONS APPLICATIONS INTEGRATING CISCO UNIFIED COMMUNICATIONS APPLICATIONS V1.0 (CAPPS) COURSE OVERVIEW: Integrating Cisco Unified Communications Applications (CAPPS) v1.0 prepares the learner for integrating Cisco Unity Connection,

More information

Reduce OPEX with easier administration as no configuration of SPAN ports is necessary Reduced CAPEX need for fewer elements at the branches

Reduce OPEX with easier administration as no configuration of SPAN ports is necessary Reduced CAPEX need for fewer elements at the branches Forked Recording Overview of Forked Recording Forked Recording (SPANless Recording) is an active recording technology, available only on Cisco CallManager platforms newer than 5.0 and selected Cisco phone

More information