Cisco Built-in-Bridge Recording Interface Configuration Guide Revision 1.1 (2014-07-01)
Table of Contents 1 Overview... 3 1.1 Purpose... 3 2 Architecture... 4 3 Requirements... 5 4 Identify Phones that Support Recording... 6 5 Configure Cisco UCM... 8 5.1 Create SIP profile for recorder... 8 5.2 Create SIP Trunk Security profile... 10 5.3 Create a SIP trunk that points to the recorder... 11 5.4 Create recording profile... 12 5.5 Create a route pattern for the recorder... 13 5.6 Enable Built-in-Bridge for all phones (optional)... 14 5.7 Configure tones for Recording... 15 5.8 Codecs configuration... 16 6 Configure phones... 18 6.1 Enable Built-in-Bridge on per-phone basis... 18 6.2 Enable recording for a line appearance... 19 7 Configure MiaRec... 20 7.1 Enable Cisco BiB interface... 20 7.2 TCP/UDP Ports for SIP Trunk Signaling Data... 20 7.3 Public IP Address... 21 7.4 UDP Ports Range for RTP Media Data... 21 8 Configure firewall... 22 Revision 1.1 2
1 Overview 1.1 Purpose This guide describes the configuration procedures required for MiaRec call recording system for interoperability with Cisco Unified Communication Manager (UCM) and phones that have Built-in- Bridge (BiB) capability. Revision 1.1 3
2 Architecture The MiaRec call recording system utilizes Built-in-Bridge call monitoring and recording capability available in 3 rd generation of Cisco phones. When call is started, Cisco UCM sends to MiaRec recording server the metadata, which contains information about recorded call such as participating device names, directory numbers, and cluster IDs. At the same time Cisco UCM instructs IP phone to fork media data to the MiaRec recording server. Central site Cisco UCM MiaRec SIP trunk TAPI link RTP audio Remote site Remote site Revision 1.1 4
3 Requirements The features utilized in this method of recording require the following: 1) 3 rd generation phones that have the built-in bridge capability (BIB) * 2) Cisco Unified Communications Manager v.8.5 or newer * - The list of phones that support the monitoring and recording features varies per version and device pack. If you have any questions, consult your Cisco account management team to determine if your telephone sets have this capability. Revision 1.1 5
4 Identify Phones that Support Recording The Cisco Unified Reporting application can be used to generate a complete list of devices that support monitoring and recording for a particular release and device pack. To do so, follow these steps: 1. Start Cisco Unified Reporting by using any of the methods that follow: a. Choosing Cisco Unified Reporting in the Navigation menu in Cisco Unified Communications Manager Administration and clicking Go. b. Choosing File > Cisco Unified Reporting at the Cisco Unified Real Time Monitoring Tool (RTMT) menu. c. Entering https://<server-name-or-ip-address>:8443/cucreports/ and then entering your authorized username and password. 2. Click System Reports in the navigation bar. 3. In the list of reports that displays in the left column, click the Unified CM Phone Feature List option. 4. Click the Generate a new report link to generate a new report, or click the Unified CM Phone Feature List link if a report already exists. 5. To generate a report of all devices that support recording, choose these settings from the respective drop-down list boxes and click the Submit button: a. Product: All b. Feature: Record 6. The List Features pane displays a list of all devices that support the recording feature. You can click on the Up and Down arrows next to the column headers (Product or Protocol) to sort the list. Revision 1.1 6
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5 Configure Cisco UCM 5.1 Create SIP profile for recorder Use the Device > Device Settings > SIP Profile menu option in Cisco Unified Communications Manager Administration to create SIP profile for recorder. The following figure illustrates creating a SIP profile for the recorder. Revision 1.1 8
You can check the Deliver Conference Bridge Identifier check box, which delivers additional information (specifically, the b-number that identifies a conference bridge) to the recorder across the SIP trunk. If the check box is left unchecked, the far-end information for the remote conference remains empty. Check the Deliver Conference Bridge Identifier check box on the remote cluster SIP profile as well. Checking this check box is not required for recording, but the conference bridge identifier helps to update the recorder when recording calls that involve a conference bridge. Revision 1.1 9
5.2 Create SIP Trunk Security profile Use the System > Security Profile > SIP Profile menu option in Cisco Unified Communications Manager Administration to create SIP Trunk Security profile for recorder. Set Incoming Transport Type to TCP+UDP. Set Outgoing Transport Type to TCP (this setting has to match the configuration of MiaRec). TCP is recommended. Uncheck option Enable Digest Authentication Set Device Security Mode parameter to Non Secure. Revision 1.1 10
5.3 Create a SIP trunk that points to the recorder Use the Device > Trunk menu option in Cisco Unified Communications Manager Administration to create SIP trunk that points to the recorder. In SIP Information section configure: Destination Address should point to ip-address or DNS name of the recorder server Destination Port should match the port on which MiaRec recorder is listening for messages from CUCM (see configuration of MiaRec below) Select the previously created SIP Trunk Security Profile for the recorder Select the previously created SIP Profile for the recorder Revision 1.1 11
5.4 Create recording profile Use the Device > Device Settings > Recording Profile menu option in Cisco Unified Communications Manager Administration to create recording profile. The following figure illustrates creating a recording profile. Set Recording Destination Address to the directory number that associates the recorder with this recording profile. The only guideline for this number: it should be possible for UCM to route it to the SIP trunk where the recorder is defined. No user is going to directly call this number, this is internal to the system. Make sure it does not collide with your numbering plan. This is why the example shows '7777'. Revision 1.1 12
5.5 Create a route pattern for the recorder Use the Call Routing > Route/Hunt > Route Pattern menu option in Cisco Unified Communications Manager Administration to create a route pattern for the recorder SIP trunk: Route Pattern should match to the Recording Destination Address in the previously created recording profile In Gateway/Route List select the SIP trunk that points to the recorder, or select a route list of which the recorder is a member Revision 1.1 13
5.6 Enable Built-in-Bridge for all phones (optional) Built-in-Bridge setting can be enabled on per-phone basis or on system level (default to all phones). Use the Access System > Service Parameters menu option in Cisco Unified Communications Manager Administration to enable Built-in-Bridge on system level. Change the following option: Clusterwide Parameters (Device - Phone) -> Built-in Bridge Enable Revision 1.1 14
5.7 Configure tones for Recording Set the service parameters for playing tone to True to allow tone to be played either to agent only, to customer only, or to both. Use the System > Service Parameters menu option in Cisco Unified Communications Manager Administration to perform the necessary configuration. Change corresponding options of group Clusterwide Parameters (Feature Call Recording). The following figure illustrates using service parameters to configure tones. Revision 1.1 15
5.8 Codecs configuration Codecs ilbc, isac, L16 and AAC-LD should be disabled for Recording-Enabled devices as they are not supported by MiaRec recording system at the moment. Use the System > Service Parameters menu option in Cisco Unified Communications Manager Administration to perform the necessary configuration. Change the following settings of group Clusterwide Parameters (System - Location and Region): ilbc Codec Enabled to Enabled for All Devices Except Recording-Enabled Devices isac Codec Enabled to Enabled for All Devices Except Recording-Enabled Devices Default Intraregion Max Audio Bit Rate to 64 kbps (G.722, G.711) If recording of conference calls is necessary, then G.722 codec should be disabled also for Recording-Enabled devices. Our product supports G.722 codec, but this codec is not supported by Cisco Conference Bridge and because of this limitation, call recording of conferenced calls is not possible. Change the following setting: G.722 Codec Enabled to Enabled for All Devices Except Recording-Enabled Devices Revision 1.1 16
Verify Region Configuration under menu System > Region. Per-region setting Max Audio Bit Rate should be set to either Use System Default or as 64 kbps (G.722, G.711) as shown on below screenshot. Revision 1.1 17
6 Configure phones 6.1 Enable Built-in-Bridge on per-phone basis Use the Device > Phone menu option in Cisco Unified Communications Manager Administration to enable Built-in-Bridge option. The following figure illustrates turning on the IP phone Built-in-Bridge to allow monitoring or recording. Revision 1.1 18
6.2 Enable recording for a line appearance Use the Device > Phone menu option in Cisco Unified Communications Manager Administration to configure line appearance of particular phone. To enable recording of an agent, set the Recording Option in the line appearance of the agent to one of the following options: o o Automatic Call Recording Enabled Selective Call Recording Enabled In Recording Profile option select the previously created recording profile from the drop-down list box The following figure illustrates enabling recording for a line appearance. Revision 1.1 19
7 Configure MiaRec Open configuration file MiaRec.ini and locate section [Protocol::CiscoBiB]. Sample configuration is shown below: ################################################################################ [Protocol::CiscoBiB] ################################################################################ # Enable Cisco Built-in-Bridge (BiB) recording #----------------------------------------------------------------- Enable=1 #----------------------------------------------------------------- # Listening port for UDP (set to 0 to disable UDP) #----------------------------------------------------------------- PortUDP=0 #----------------------------------------------------------------- # Listening port for TCP (set to 0 to disable TCP) #----------------------------------------------------------------- PortTCP=5070 #----------------------------------------------------------------- # Public IP-address, if MiaRec is behind NAT # The NAT router should be configured for port forwarding #----------------------------------------------------------------- PublicIp= #----------------------------------------------------------------- # Ports to use for RTP (min, max) #----------------------------------------------------------------- PortRtpMin = 20000 PortRtpMax = 21999 #----------------------------------------------------------------- Restart of MiaRec service is required when configuration file is modified. 7.1 Enable Cisco BiB interface Set parameter Enable to yes or 1: Enable = yes 7.2 TCP/UDP Ports for SIP Trunk Signaling Data Change parameters PortUDP and PortTCP according to the port configuration in section 5.3 Create a SIP trunk that points to the recorder. In the following example MiaRec will listen on port 5070 for both UDP and TCP protocols. PortUDP = 5070 PortTCP = 5070 Revision 1.1 20
7.3 Public IP Address Specify public IP address if MiaRec server is located behind NAT: PublicIp = 1.2.3.4 Make sure that port forwarding is configured properly on your NAT router. If MiaRec server is not behind NAT, then leave this parameter empty: PublicIp = 7.4 UDP Ports Range for RTP Media Data If necessary, change default values of UDP port range for RTP media packets: PortRtpMin = 20000 PortRtpMax = 21999 Make sure that the port range is large enough for anticipated number of concurrently recorded calls. One concurrent call requires two UDP ports for receiving media streams from agent s phone. Revision 1.1 21
8 Configure firewall If firewall is used on MiaRec recording server, then add exclusion rules for ports mentioned in sections: 7.2 TCP/UDP Ports for SIP Trunk Signaling Data 7.4 UDP Ports Range for RTP Media Data Revision 1.1 22