Cisco Unified Communications Manager (CallManager)

Document ID: 82055

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This series of interactive documents uses a typical head office and branch office in order to demonstrate some common voice network configurations. This interactive diagram allows you to view network configurations:

This image is interactive! Click a component to view configuration and troubleshooting information.

Refer to Unified CallManager Interactive Voice Network Configuration and Troubleshooting Case Study for requirements assumed by this series of interactive documents.

Configure

For the San Jose Site, we will utilize the default Software Conference bridge. We will, however, configure a Transcoder in Hardware.

For the RTP site, this configuration example configures a local hardware conference bridge and a transcoder in order to ensure that local resources are utilized instead of WAN resources.

1. Configure a Transcoder on the San Jose Router
2. Configure a Conference Bridge on the RTP router
3. Configure a Transcoder on the RTP router
4. Configure Multicast Music-On-Hold in Unified CallManager
5. Ensure Multicast Routing is Enabled
6. Assign Media Resources to the SanJose and RTP MRG and MRGL

Configure a Transcoder on the San Jose Router

In order to configure a transcoder on the San Jose router, complete these steps:

1. Click Service > Media Resource > Transcoder > Add a New Transcoder.
   
   
2. For the transcoder type, choose Cisco IOS Enhanced Media Termination Point (because this example uses an NM-HDV2 module).
   
   
   
   
3. Use the configuration:

   hostname SanJose
   
   voice-card 2
   dspfarm
   dsp services dspfarm
   
   sccp local Vlan120
   sccp ccm 171.69.103.225 identifier 1
   sccp
   !
   sccp ccm group 1
   associate ccm 1 priority 1
   associate profile 1 register MTP12345SANJOSE !--- MTP12345SANJOSE matches the
                                                !--- CallManager configuration for the Transcoder
   !
   dspfarm profile 1 transcode
   codec g711ulaw
   codec g711alaw
   codec g729ar8
   codec g729abr8
   codec gsmfr
   maximum sessions 2
   associate application SCCP
   !

Configure a Conference Bridge on the RTP Router

In order to configure a conference bridge on the RTP router, complete these steps:

1. Click Service > Media Resource > Conference Bridge >Add a New Conference Bridge.
   
   
2. For the conference bridge type, choose Cisco IOS Enhanced Conference Bridge (because this example uses an NM-HDV2 module).
   
   
   
   
3. Use this configuration:

   hostname RTP
   !
   voice-card 1
   dspfarm
   dsp services dspfarm
   !
   
   sccp local Vlan110
   sccp ccm 171.69.103.225 identifier 1
   sccp
   !
   sccp ccm group 1
   associate ccm 1 priority 1
   associate profile 1 register CFB123456789123! !--- CFB123456789123 matches the 
                                                 !--- CallManager configuration for the Conference Bridge
   !
   dspfarm profile 1 conference
   codec g711ulaw
   codec g729r8
   maximum sessions 1
   associate application SCCP
   !

Configure a Transcoder on the RTP Router

In order to configure a transcoder on the RTP router, complete these steps:

1. Click Service > Media Resource > Transcoder > Add a New Transcoder.
   
   
2. For the transcoder type, choose Cisco IOS Enhanced Media Termination Point (because this example uses an NM-HDV2 module).
   
   
   
   
3. Use this configuration:

   hostname RTP
   !
   voice-card 1
   dspfarm
   dsp services dspfarm
   !
   
   sccp local Vlan110
   sccp ccm 171.69.103.225 identifier 1
   sccp
   !
   sccp ccm group 1
   associate ccm 1 priority 1
   associate profile 2 register MTP123456789123 !--- MTP123456789123 matches the 
                                                !--- CallManager configuration for the Transcoder
   associate profile 1 register CFB123456789123
   !
   dspfarm profile 2 transcode
   codec g711ulaw
   codec g711alaw
   codec g729ar8
   codec g729abr8
   codec gsmfr
   maximum sessions 1
   associate application SCCP
   !

Configure Multicast Music-On-Hold in Unified CallManager

In order to configure Music-On-Hold (MOH), complete these steps:

1. Click Service > Media Resource > Music On Hold Audio Source.
   
   This page allows you to configure the audio source and enable Multicast:
   
   
   
   
2. Configure the MOH server as shown in this image:
   
   
   
   Note: This example configures multicast MOH and IP address in order to increment multicast on. This example also extends the max hop count to three. For more information on different configuration parameters and best practices, refer to the Music on Hold section in Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x.
   
   
3. By default, MOH is transmitted using G.711 ulaw codec. In order to generate G.729 streams for bandwidth efficiency, click Service > Service Parameters > Cisco IP Voice Media Streaming App in order to enable the G.729 codec in the Cisco IP voice media streaming application.
   
   
   
   Note: To select multiple codecs, press the Ctrl key and click the desired codec.
   
   For more information on best practices, refer to the Music on Hold section of Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x.

Ensure Multicast Routing is Enabled

Multicast Routing is beyond the scope of this document. However, at a minimum, you must enable the multicast-routing global command and the ip pim sparse-dense-mode command on the necessary interfaces that are used in the multicast stream.

For more information on multicast routing, refer to the Cisco Support website.

Assign Media Resources to the San Jose and RTP MRG and MRGL

In order to ensure that the San Jose and RTP locations use the correct media resources, you must configure a media resource group (MRG) to include to appropriate resources, and then assign that MRG to a resource group list (MRGL) that corresponds to that location.

For example, create an MRG that includes San Jose resources, and assign that MRG to an MRGL that belongs to the San Jose location.

This configuration ensures that media resources are used locally, instead of the WAN that uses scarce bandwidth.

In order to assign media resources to the San Jose and RTP MRG and MRGL, complete these steps:

1. Configure the San Jose and RTP MRGs:
   
   
   1. Click Service > Media Resource > Media Resource Group > Add a New Media Resource Group.
      
      
      
      
   2. Assign only San Jose media resources to the the MRG-SJ (include the MOH server), and verify that the Use Multicast check box is checked. Otherwise, multicast MOH will not work.
      
      
      
      
   3. Assign only RTP media resources to the the MRG-RTP (include MOH server), and verify that the Use Multicast check box is checked. Otherwise, multicast MOH will not work.
      
      
2. Create the MRGLs for San Jose and RTP, and assign the proper MRG:
   
   
   1. Click Service > Media Resource > Media Resource Group List > Add a New Media Resource Group List.
      
      
      
      
   2. Assign the San Jose MRG (MRG-SJ) to the San Jose MRGL (MRGL-SJ).

      

   3. Assign the RTP MRG (MRG-RTP) to the RTP MRGL (MRGL-RTP).

Verify

Before you verify this configuration, you must assign the media resource group lists (MRGLs) to the IP phones in the phone configurations. Refer to these documents for more information:

• Phone Configuration (San Jose) Interactive Voice Network Configuration Example
• Phone Configuration (RTP) Interactive Voice Network Configuration Example

In order to verify that devices use their assigned resources, conference three IP phones across the two sites to make sure they use G.711 and G.729. For example, in RTP call and RTP phone then conference to a San Jose phone, and execute the show dspfarm dsp active command.

RTP#show dspfarm dsp active
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

1 5 1.0.6 UP 1 USED conf 1 6 45559 45554
1 5 1.0.6 UP 1 USED conf 1 7 45559 45557
1 5 1.0.6 UP 1 USED conf 1 8 44959 44949

Total number of DSPFARM DSP channel(s) 1

This output shows an active conference between 3 phones. The initiator is an IP Phone at the RTP site. Notice the three (3) active DSP sessions on the RTP router.