MoH Scenarios

A. Unicast G711 everywhere

1. Create region for MoH. G711 with all other regions.

2. Do not enable multicast in audio source, server, MRG

3. Enable only G711 in service parameter (Default)

B. G711 multicast everywhere

1. Create region for MoH. G711 with all other regions.

2. Enable multicast in audio source, server, MRG. Set hop count to 5.

3. Enable only G711 in service parameter (Default)
4. Enable multicast in infrastructure (igmp snooping in switches, pim sparse-dense in routers)

C. G711 multicast everywhere, remote site uses MoH from local SRST flash

1. Create region for MoH. G711 with all other regions.

2. Enable multicast in audio source, server, MRG. Set hop count to 1.

3. Enable only G711 in service parameter (Default)
4. Enable multicast in HQ and Remote LAN infrastructure (igmp snooping in switches, pim sparse-dense in routers), Disable multicast in WAN infrastructure.

5. Enable SRST MOH feature from flash.

6. Enable loopback interface in SRST router for PSTN users to hear MoH.

D. G711 multicast at HQ, G711 unicast at Remote

1. Create region for MoH. G711 with all other regions.

2. Enable multicast in audio source, server, MRG. Set hop count to 1.

3. Enable only G711 in service parameter (Default)
4. Enable multicast in HQ and Remote LAN infrastructure (igmp snooping in switches, pim sparse-dense in routers), Disable multicast in WAN infrastructure.

5. Create two MRG, one with multicast enabled for HQ and one with no mulitcast for BR1.

E. G711 unicast at HQ site, G729 unicast at remote

1. Enable Service parameter for G711 and G729

2. Use the HQ device pool for MOH server (assuming HQ device pool uses g729 for remote)

3. Do not enable multicast

F. G711 multicast at HQ, g729 multicast at remote

1. Enable Service parameter for G711 and G729

2. Use the HQ device pool for MOH server (assuming HQ device pool uses g729 for remote)

3. Enable multicast in audio source, server, MRG

4. Enable multicast in infrastructure

G. g729 unicast at remote using transcode

1. Enable only G711 in service parameter

2. Put MoH server in HQ device pool (assuming HQ uses G729 for remote)

3. Add a transcoder to MRGL of MoH server

I. HQ uses G711 multicast, remote uses G729 unicast, all PSTN uses hear tone on hold

1. Use HQ device pool for MoH server

2. Enable multicast on Audio source and server.

3. Create 2 MRG, one with multicast, one without multicast

4. Enable g711 and g729 in the service parameter

5. Create a third MRG and MRGL with no MOH and apply that directly to gateways

MoH server using Trancoder

If the remote device does not support the codec of the MOH stream, then MOH server can trigger a transcoder. You would need to configure a MRGL with transcoder resource in the MoH servers device pool. In this case, CCM does not use the trancoder of the remote device. It will only use the transcoder of MOH server.

MRGL for gateways

Make sure you always specify MRGLs for Gateways. They dont default to the device pool to get the MRGL. (unlike phones, where if you set the MRGL to none, they default to the device pool)

MoH Folders

1. Folder to drop new audio files

C:\Program Files\Cisco\MOH\DropMOHAudioSourceFilesHere

2. Folder were all converted music files are stored

C:\Program Files\Cisco\TFTPPath\MOH

3. After you add the music on hold audio stream in Callmanager admin, it will move the audio files to

C:\Program Files\Cisco\TFTPPath and

C:\Program Files\Cisco\MOH

MOH settings for Multicast

a. Enable Multicast on audio source (Play cont. also)

b. Enable Multicast on MOH audio server, Increment by ip address rather than port, set hops to greater than number of hops required to reach the phone.

c. Enable Multicast on MRG group.

d. To receive g729 MOH, set service parameter in IP Voice media service and add g729 annex A setting

e. Enable these commands on the following equipment

  i. Wan routers :

           ip multicast-routing

           int fa0/0.2

           ip pim sparse-dense-mode

    ii. 6500 switch – enable igmp (main site)

             set igmp enable

    iii. 3550 switch (remote site)

             ip igmp snooping vlan <voicevlan>

Software Conference

As per this link

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_configuration_example09186a0080114c64.shtml 
Software conference devices only support G.711 codecs. Cisco CallManager 4.0 and later also support G.729 codecs. Still, the default codec is G.711. However, if the Meet-Me conference is initiated by a phone that has a G.711 codec, G.729 devices are not able to join the meeting unless there is a transcoder involved. A transcoder is needed to handle the translation from one codec to the other one. A similar issue arises with Ad-Hoc conferences. If the phone that initiates the conference tries to include a phone with a different codec, it fails. An error message is displayed on the traces. Hardware conference devices can provide transcoding for G.711, G.729, and G.723 codecs.

This did not work in the lab. After changing the region to use G729 and trying to do a conference, getting a fast busy. Also event viewer is reporting the error

Error: MtpNoMoreResourcesAvailable – No more MTP resources available.
App ID: Cisco CallManager
Cluster ID: StandAloneCluster
Node ID: 10.1.1.1

NM-HDV transcoding and conferencing

* NM-hdv has 5 PVDM SIMM slots

* Each pvdm has 3 dsps

* Each dsp can handle 4 calls (g729,g711)

* Each dsp can handle 4 transcoding sessions (g729,g711)

* If using GSM, each dsp can handle 3 transcoding sessions

* Each dsp can handle 1 conference session.

* max capacity for transcoding (g729,g711) on a NM-HDV is (4 per DSP) * 3 DSPs per SIMM * 5 SIMMs per module = 60 sessions.

* max capacity for conferencing on a NM-HDV is (1 per DSP) * 3 DSPs per SIMM * 5 SIMMs per module = 15 sessions.

CUE – CME – CCM Integration

When CME and CUE are integrated and you place a call from a CCM to CME using G729 and the call rolls over to CUE, then you need to have transcoding enabled in CME for this to work. This is needed because CME is controlling the call leg to CUE via SIP. There is no point in having transcoder at CCM and this will not be used.

Issues with using MTP in gateways

Enabled MTP in the H323 gateway. Placed an outgoing call using that gateway. From the IP Phone, none of the features like Hold, transfer, conference is working.

MMOH using router

1. MMOH from the MOH server to IP Phones at remote site

Enable IP Multicasting on all routers

ip multicast-routing

interface fastethernet 0/0

ip pim sparse-dense-mode

2. MMOH from the remote site gateway to IP Phones at remote site

Enable MMOH in SRST mode

3. MMHOH from the MOH server to PSTN users coming via the remote gateway

ccm-manager music-on-hold

Enable IP Multicasting on all routers
voice-class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

dial-peer voice 10 voip

incoming called-number .

voice-class codec 1

4. MMOH from the remote site gateway to PSTN users when the gateway is in H323 mode

ccm-manager music-on-hold

interface loopback 0

ip address 10.1.1.10 255.255.255.255

call-manager-fallback

multicast moh 239.1.1.1 port 16384 route 10.1.1.10

voice-class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

dial-peer voice 10 voip

incoming called-number .

voice-class codec 1

MMoh via SRST in IOS

Prerequisites 

* SRST router can play MOH file as a multicast to users in the remote site. Unicast MOH is not supported. This saves valuable bandwidth required for streaming MOH over wan.

* The codec used for MOH to remote site (between MOH server's region and Site2) should be g711.

* Only a single MOH source should be used across all phones for this feature to work. (User and Network hold should be same)

* Prevent MOH from reaching branch sites or even crossing wan. Set MMOH hop count to 1 in callmanager.

* Enter a base multicast address (say 239.1.1.1, 16384) in Callmanager. This will be used for g711ulaw. 239.1.1.2 will be used for g711alaw, 239.1.1.3 for g729 and 239.1.1.4 for wideband (as long all these codecs are checked in service parameter and increment by ip address is selected)

Configuration on SRST router

* ccm-manager music-on-hold —- enable multicast moh on ios router

*interface loopback 0, ip address ##1.1.1.1## ##255.255.255.255##  — Configure this so that Pots users will hear multicast moh.

* Configure SRST in the router

    – call-manager-fallback

    – ip source-address ##fastE ip address## port 2000

    – max-ephones , max-dn

    –  moh ##filename## (specifies the moh file – .au or .wav, 8khz, mono)

    – multicast moh 239.1.1.1 port 16384 route ##ipaddress-list##  (if route option is not used, it will use ipaddress used in "ip source-address" command.

    for ex: multicast moh 239.1.1.1 port 16384 route ##fastEipaddress## ##loopbackipaddress##

Enable MOH with g729

Set the MOH service parameter for G729 and G711.

Put MOH server in same region as Site1

Use g729 between Site 1 and Site2/3.

Enable MMOH

a. Enable Multicast option on MOH server. Use increment by IP Address rather than port. (239.x.x.x). Set the hop counts according to how many router hops the packet has to traverse. Also dont forget to enable PIM on routers.

ip multicast-routing

int s0

ip pim sparse-dense-mode

b. Enable multicast option on audio source

c. Enable Multicast option on MRGroup.

d. Apply this group to a MRGList

e. Apply the list to device pools.

f. Reset moh servers.

IOS Transcoding and Conferencing on NM-HDV

1. Copy mac-address of the interface that you want to use to register with Callmanager. (match with sccp local command)

2. Add a new IOS conference bridge with device name "CFBxxxxxxxxxxxx" or a new IOS Transcoder with "MTPxxxxxxxxxxxx".

3. Type the following commands in the router

########Enable DSPfarm services under appropriate slot################

voice-card 1
 dsp services dspfarm

###############Activates dsp farm and sets max number of sessions#################

dspfarm transcoder maximum sessions 4   —- configures max number of conf. sessions (4 transcoding sessions take 1 dsp)
dspfarm confbridge maximum sessions 2  —- configurs max number of transcoding sessions (1 conferencing session take 1 dsp)

dspfarm

sccp local FastEthernet0/0.2  —- specifies a local interface to use to register with callmanager. Use mac-address of this interface.
sccp  —- globally enables SCCP.
sccp ccm 10.1.21.33 priority 1  —- specifies the callmanager with priority of 1. Specify up to 4 callmanagers.

sccp ccm 10.1.21.34 priority 2 —– specify backup callmanager

If the transcoder or Cfb does not register with callmanager,

a. try reset it from callmanager.

b. do a "no sccp" and then do a "sccp" and see if it registers.

Output of "sh sccp"
SITE3-R3#sh sccp
SCCP Admin State: UP
Gateway IP Address: 192.168.221.240
Switchover Method: IMMEDIATE, Switchback Method: GUARD_TIMER
Switchback Guard Timer: 1200 sec, IP Precedence: 5
Max Supported MTP sessions: 0
User Masked Codec list: None
Transcoding Oper State: ACTIVE – Cause Code: NONE
Active Call Manager: 10.1.21.33, Port Number: 2000
TCP Link Status: CONNECTED
Conferencing Oper State: ACTIVE – Cause Code: NONE
Active Call Manager: 10.1.21.33, Port Number: 2000
TCP Link Status: CONNECTED
Call Manager: 10.1.21.33, Port Number: 2000
                Priority: 1, Version: 3.1 or Higher 

Meet me – Issue with using a range

When specifying a range for Meet-me, it gives a fast busy if the user tries to dial a meet me DN that is in use. Say, if the meet me range is 79XX and 7900 is already in use and a user tries to call 7900 it gives a fast busy. The user must know upfront which meet me number is available.