Media Resources Architecture
To properly design the media resource allocation strategy for an enterprise, it is critical to understand the Cisco Unified CM architecture for the various media resource components. The following sections highlight the important characteristics of media resource design with Unified CM.
Media Resource Manager
The Media Resource Manager (MRM), a software component in the Unified CM, determines whether a media resource needs to be allocated and inserted in the media path. This media resource may be provided by the Unified CM IP Voice Media Streaming Application service or by digital signal processor (DSP) cards. When the MRM decides and identifies the type of the media resource, it searches through the available resources according to the configuration settings of the media resource group list (MRGL) and media resource groups (MRGs) associated with the devices in question. MRGLs and MRGs are constructs that hold related groups of media resources together for allocation purposes and are described in detail in the section on Media Resource Groups and Lists.
Figure 7-1 shows how a media resource such as a transcoder may be placed in the media path between an IP phone and a Cisco Unified Border Element when a common codec between the two is not available.
Figure 7-1 Use of a Transcoder Where a Common Codec Is Not Available
Unified CM communicates with media resources using Skinny Client Control Protocol (SCCP). This messaging is independent of the protocol that might be in use between Unified CM and the communicating entities. Figure 7-2 shows an example of the message flow, but it does not show all of the SCCP or SIP messages exchanged between the entities.
Figure 7-2 Message Flow Between Components
Cisco IP Voice Media Streaming Application
The Cisco IP Voice Media Streaming Application provides the following software-based media resources:
Music on Hold (MoH)
Media termination point (MTP)
When the IP Voice Media Streaming Application is activated, one of each of the above resources is automatically configured. Conferencing, annunciator, and MTP services can be disabled if required. If these resources are not needed, Cisco recommends that you disable them by modifying the appropriate service parameter in the Unified CM configuration. The service parameters have default settings for the maximum number of connections that each service can handle. For details on how to modify the service parameters, refer to the appropriate version of the
Cisco Unified Communications Manager Administration Guide
, available at
Give careful consideration to situations that require multiple resources and to the load they place on the IP Voice Media Streaming Application. The media resources can reside on the same server as Unified CM or on a dedicated server not running the Unified CM call processing service. If your deployment requires more than the default number of any resource, Cisco recommends that you configure that resource to run on its own dedicated server. Cisco strongly recommends that you do not activate the Cisco IP Voice Streaming Media Application on a Cisco Unified CM node that has a high call processing load because it can adversely affect the performance of Cisco Unified CM. If heavy use of media resources is expected within a deployment, Cisco recommends deploying dedicated Unified CM media resource nodes (non-publisher nodes that do not perform call processing within the cluster) or relying on hardware-based media resources. Software-based media resources on Unified CM nodes are intended for small deployments or deployments where need for media resources is limited.
Voice termination applies to a call that has two call legs, one leg on a time-division multiplexing (TDM) interface and the second leg on a Voice over IP (VoIP) connection. The TDM leg must be terminated by hardware that performs encoding/decoding and packetization of the stream. This termination function is performed by a digital signal processor (DSP) resource residing in the same hardware module, blade, or platform.
All DSP hardware on Cisco TDM gateways is capable of terminating voice streams, and certain hardware is also capable of performing other media resource functions such as conferencing or transcoding (see Transcoding and Transcoding). The DSP hardware has either fixed DSP resources that cannot be upgraded or changed, or modular DSP resources that can be upgraded.
The number of supported calls per DSP depends on the computational complexity of the codec used for a call and also on the complexity mode configured on the DSP. Cisco IOS enables you to configure a complexity mode on the hardware module. Hardware platforms such as the PVDM2 and PVDM3 DSPs support three complexity modes: medium, high and flex mode. Some of the other hardware platforms support only medium and high complexity modes.
Medium and High Complexity Mode
You can configure each DSP separately as either medium complexity, high complexity, or flex mode (PVDM3 DSPs and those based on C5510). The DSP treats all calls according to its configured complexity, regardless of the actual complexity of the codec of the call. A resource with configured complexity equal or higher than the actual complexity of the incoming call must be available, or the call will fail. For example, if a call requires a high-complexity codec but the DSP resource is configured for medium complexity mode, the call will fail. However, if a medium-complexity call is attempted on a DSP configured for high complexity mode, then the call will succeed and Cisco IOS will allocate a high-complexity mode resource.
Flex mode, available on hardware platforms that use the C5510 chipset and on PVDM3 DSPs, eliminates the requirement to specify the codec complexity at configuration time. A DSP in flex mode accepts a call of any supported codec type, as long as it has available processing power.
For C5510-based DSPs, the overhead of each call is tracked dynamically via a calculation of processing power in millions of instructions per second (MIPS). Cisco IOS performs a MIPS calculation for each call received and subtracts MIPS credits from its budget whenever a new call is initiated. The number of MIPS consumed by a call depends on the codec of the call. The DSP will allow a new call as long as it has remaining MIPS credits greater than or equal to the MIPS required for the incoming call.
Similarly, PVDM3 DSP modules use a credit-based system. Each module is assigned a fixed number of "credits" that represent a measure of its capacity to process media streams. Each media operation, such as voice termination, transcoding, and so forth, is assigned a cost in terms of credits. As DSP resources are allocated for a media processing function, its cost value is subtracted from the available credits. A DSP module runs out of capacity when the available credits run out and are no longer sufficient for the requested operation. The credit allocation rules for PVDM3 DSPs are rather complex.
For proper DSP sizing of Cisco ISR gateways, you can use the Cisco Unified Communications Sizing Tool (Unified CST), available to Cisco employees and partners at
. If you are not a Cisco partner, you can use the DSP Calculator at
. For other Cisco non-ISR gateway platforms (such as the Cisco 1700, 2600, 3700, and AS5000 Series) and/or Cisco IOS releases preceding and up to 12.4 mainline, you can access the legacy DSP calculator at
Flex mode has an advantage when calls of multiple codecs must be supported on the same hardware because flex mode can support more calls than when the DSPs are configured as medium or high complexity. However, flex mode does allow oversubscription of the resources, which introduces the risk of call failure if all resources are used. With flex mode it is possible to have fewer DSP resources than with physical TDM interfaces.
Compared to medium or high complexity mode, flex mode has the advantage of supporting the most G.711 calls per DSP. For example, a PVDM2-16 DSP can support 8 G.711 calls in medium complexity mode or 16 G.711 calls in flex mode.
A transcoder is a device that converts an input stream from one codec into an output stream that uses a different codec. Starting with Cisco IOS Release 15.0.1M, a transcoder also supports transrating, whereby it connects two streams that utilize the same codec but with a different packet size.
Transcoding from G.711 to any other codec is referred to as traditional transcoding. Transcoding between any two non-G.711 codecs is called universal transcoding and requires Universal Cisco IOS transcoders. Universal transcoding is supported starting with Cisco IOS Release 12.4.20T. Universal transcoding has a lower DSP density than traditional transcoding.
In a Unified CM system, the typical use of a transcoder is to convert between a G.711 voice stream and the low bit-rate compressed voice stream G729a. The following cases determine when transcoder resources are needed:
Single codec for the entire system
A single codec is generally used in a single-site deployment that usually has no need for conserving bandwidth. When a single codec is configured for all calls in the system, then no transcoder resources are required. In this scenario, G.711 is the most common choice that is supported by all vendors.
Multiple codecs in use in the system, with all endpoints capable of all codec types
The most common reason for multiple codecs is to use G.711 for LAN calls to maximize the call quality and to use a low-bandwidth codec to maximize bandwidth efficiency for calls that traverse a WAN with limited bandwidth. Cisco recommends using G.729a as the low-bandwidth codec because it is supported on all Cisco Unified IP Phone models as well as most other Cisco Unified Communications devices, therefore it can eliminate the need for transcoding. Although Unified CM allows configuration of other low-bandwidth codecs between regions, some phone models do not support those codecs and therefore would require transcoders. They would require one transcoder for a call to a gateway and two transcoders if the call is to another IP phone. The use of transcoders is avoided if all devices support and are configured for both G.711 and G.729 because the devices will use the appropriate codec on a call-by-call basis.
Multiple codecs in use in the system, and some endpoints support or are configured for G.711 only
This condition exists when G.729a is used in the system but there are devices that do not support this codec, or a device with G.729a support may be configured to not use it. In this case, a transcoder is also required. Devices from some third-party vendors may not support G.729.
A transcoder is also capable of performing the same functionality as a media termination point (MTP). In cases where transcoder functionality and MTP functionality are both needed, a transcoder is allocated by the system. If MTP functionality is required, Unified CM will allocate either a transcoder or an MTP from the resource pool, and the choice of resource will be determined by the media resource groups, as described in the section on Media Resource Groups and Lists.
To finalize the design, it is necessary to know how many transcoders are needed and where they will be placed. For a multi-site deployment, Cisco recommends placing a transcoder local at each site where it might be required. If multiple codecs are needed, it is necessary to know how many endpoints do not support all codecs, where those endpoints are located, what other groups will be accessing those resources, how many maximum simultaneous calls these device must support, and where those resources are located in the network.
Audio Transcoding Resources
DSP resources are required to perform transcoding. Those DSP resources can be located in the voice modules and the hardware platforms for transcoding that are listed in the following sections.
Hardware Transcoder (Cisco NM-HDV2, NM-HD-1V/2V/2VE, and PVDM2 DSPs)
The number of sessions supported on each DSP is determined by the codecs used in universal transcoding mode. The following guidelines and considerations apply to these DSP resources:
Transcoding is available between G.711 mu-law or a-law and G.729a, G.729ab, G.722, and iLBC. A single PVDM2-16 can support 8 sessions for transcoding between low and medium complexity codecs (such as G.711 and G.729a or G.722) or 6 sessions for transcoding between low and high complexity codecs (such as G.711 and G.729 or iLBC).
Note If transcoding is not required between G.711 and G.722, Cisco recommends that you do not include G.722 in the Cisco IOS configuration of the dspfarm profile. This is to preclude Unified CM from selecting G.722 as the codec for a call in which transcoding is required. DSP resources configured as Universal Transcoders are required for transcoding between G.722 and other codecs.
Cisco Unified IP Phones use only the G.729a variants of the G.729 codecs. The default for a new DSP farm profile is G.729a/G.729ab/G.711u/G.711a. Because a single DSP can provide only one function at a time, the maximum sessions configured on the profile should be specified in multiples of 8 to prevent wasted resources.
For capacity information on PVDM2 modules, refer to the
High-Density Packet Voice Digital Signal Processor Module for Cisco Unified Communications Solutions
data sheet, available at
Hardware Transcoder (Cisco WS-SVC-CMM-ACT)
The following guidelines and considerations apply to this DSP resource:
Transcoding is available between G.711 mu-law or a-law and G.729, G.729b, or G.723.
There are 4 DSPs per ACT that may be allocated individually to DSP pools.
The CCM-ACT can have 16 transcoded calls per DSP or 64 per ACT. The ACT reports resources as streams rather than calls, and a single transcoded call consists of two streams.
Hardware Transcoder (Cisco NM-HDV and 1700 Series Routers)
The following guidelines and considerations apply to these DSP resources:
This hardware utilizes the PVDM-256K type modules, and each DSP provides 2 transcoding sessions.
The NM-HDV may have up to 4 PVDM-256K modules, and the Cisco 1700 Series Routers may have 1 or 2 PVDM-256K modules. The Cisco 1751 Router has a chassis limit of 16 sessions, and the Cisco 1760 Router has a chassis limit of 20 sessions.
NM-HDV and NM-HDV2 modules may be used simultaneously in a single chassis for voice termination but may not be used simultaneously for other media resource functionality. Only one type of DSP farm configuration may be active at one time (either the NM-HDV or the HM-HDV2) for conferencing, MTP, or transcoding.
Transcoding is supported from G.711 mu-law or a-law to any of G.729, G.729a, G.729b, or G.729ab codecs.
Hardware Transcoder (PVDM3 DSP)
PVDM3 DSPs are hosted by Cisco 2900 Series and 3900 Series Integrated Services Routers. They support only non-secure transcoding with Unified CM, however secure transcoding is supported with Unified CME. As with voice termination and conferencing, each transcoding session debits the available credits for each type of PVDM3 DSPs. The available credits determine the total capacity of the DSP.
For example, a PVDM3-16 can support 12 sessions for transcoding between low and medium complexity codecs (such as G.711 and G.729a or G.722) or 10 sessions for transcoding between low and high complexity codecs (such as G.711 and G.729 or iLBC).
For capacity information on PVDM3 modules, refer to the
High-Density Packet Voice Video Digital Signal Processor Module for Cisco Unified Communications Solutions
data sheet, available at
Video interoperability is the audio and video support for point-to-point calls between Cisco TelePresence System (CTS) endpoints, other Cisco Unified Communications video endpoints, and third-party video endpoints. Prior to Cisco Unified CM 8.5, video interoperability between the various families of video endpoints was possible only with the insertion of a video component between endpoints, such as a video transcoder or a multipoint control unit (MCU).
Cisco Unified CM 8.5 and later releases not only offer native video interoperability between the various video endpoint family types, point-to-point, but also provide better video interoperability in general with H.264 codec negotiation in SIP and H.323 protocols and enable the endpoints to negotiate high definition (HD) resolutions when available. Video interoperability, however, is dependent on the endpoints to support the interoperation. For further information, refer to
Interoperability Between CTS Endpoints and Other Cisco Endpoints or Devices
, available at
Media Termination Point (MTP)
A media termination point (MTP) is an entity that accepts two full-duplex media streams. It bridges the streams together and allows them to be set up and torn down independently. The streaming data received from the input stream on one connection is passed to the output stream on the other connection, and vice versa. MTPs have many possible uses, such as:
– DTMF Relay over SIP Trunks
– H.323 Supplementary Services
– H.323 Outbound Fast Connect
Re-Packetization of a Stream
An MTP can be used to transcode G.711 a-law audio packets to G.711 mu-law packets and vice versa, or it can be used to bridge two connections that utilize different packetization periods (different sample sizes). Note that re-packetization requires DSP resources in a Cisco IOS MTP.
DTMF tones are used during a call to signal to a far-end device for purposes of navigating a menu system, entering data, or other manipulation. They are processed differently than DTMF tones sent during a call setup as part of the call control. There are several methods for sending DTMF over IP, and two communicating endpoints might not support a common procedure. In these cases, Unified CM may dynamically insert an MTP in the media path to convert DTMF signals from one endpoint to the other. Unfortunately, this method does not scale because one MTP resource is required for each such call. The following sections help determine the optimum amount of MTP resources required, based on the combination of endpoints, trunks, and gateways in the system.
If Unified CM determines that an MTP needs to be inserted but no MTP resources are available, it uses the setting of the service parameter
Fail call if MTP allocation fails
to decide whether or not to allow the call to proceed.
DTMF Relay Between Endpoints
The following methods are used to relay DTMF from one endpoint to another.
Named Telephony Events (RFC 2833)
Named Telephony Events (NTEs) defined by RFC 2833 are a method of sending DTMF from one endpoint to another after the call media has been established. The tones are sent as packet data using the already established RTP stream and are distinguished from the audio by the RTP payload type field. For example, the audio of a call can be sent on a session with an RTP payload type that identifies it as G.711 data, and the DTMF packets are sent with an RTP payload type that identifies them as NTEs. The consumer of the stream utilizes the G.711 packets and the NTE packets separately.
Key Press Markup Language (RFC 4730)
The Key Press Markup Language (KPML) is defined in RFC 4730. Unlike NTEs, which is an in-band method of sending DTMF, KPML uses the signaling channel (out-of-band, or OOB) to send SIP messages containing the DTMF digits.
KPML procedures use a SIP SUBSCRIBE message to register for DTMF digits. The digits themselves are delivered in NOTIFY messages containing an XML encoded body.
Unsolicited Notify (UN)
Unsolicited Notify procedures are used primarily by Cisco IOS SIP Gateways to transport DTMF digits using SIP NOTIFY messages. Unlike KPML, these NOTIFY messages are unsolicited, and there is no prior registration to receive these messages using a SIP SUBSCRIBE message. But like KPML, Unsolicited Notify messages are out-of-band.
Also unlike KPML, which has an XML encoded body, the message body in these NOTIFY messages is a 10-character encoded digit, volume, and duration, describing the DTMF event.
H.245 Signal, H.245 Alphanumeric
H.245 is the media control protocol used in H.323 networks. In addition to its use in negotiating media characteristics, H.245 also provides a channel for DTMF transport. H.245 utilizes the signaling channel and, hence, provides an out-of-band (OOB) way to send DTMF digits. The Signal method carries more information about the DTMF event (such as its actual duration) than does Alphanumeric.
Cisco Proprietary RTP
This method sends DTMF digits in-band, that is, in the same stream as RTP packets. However, the DTMF packets are encoded differently than the media packets and use a different payload type. This method is not supported by Unified CM but is supported on Cisco IOS Gateways.
Skinny Client Control Protocol (SCCP)
The Skinny Client Control Protocol is used by Unified CM for controlling the various SCCP-based devices registered to it. SCCP defines out-of-band messages that transport DTMF digits between Unified CM and the controlled device.
DTMF Relay Between Endpoints in the Same Unified CM Cluster
The following rules apply to endpoints registered to Unified CM servers in the same cluster:
Calls between two non-SIP endpoints do not require MTPs.
All Cisco Unified Communications endpoints other than SIP send DTMF to Unified CM via various signaling paths, and Unified CM forwards the DTMF between dissimilar endpoints. For example, an IP phone may use SCCP messages to Unified CM to send DTMF, which then gets sent to an H.323 gateway via H.245 signaling events. Unified CM provides the DTMF forwarding between different signaling types.
Calls between two Cisco SIP endpoints do not require MTPs.
All Cisco SIP endpoints support NTE, so DTMF is sent directly between endpoints and no conversion is required.
A combination of a SIP endpoint and a non-SIP endpoint might require MTPs.
To determine the support for NTE in your devices, refer to the product documentation for those devices. Support of NTE is not limited to SIP and can be supported in devices with other call control protocols. Unified CM has the ability to allocate MTPs dynamically on a call-by-call basis, based on the capabilities of the pair of endpoints.
DTMF Relay over SIP Trunks
A SIP trunk configuration is used to set up communication with a SIP User Agent such as another Cisco Unified CM cluster or a SIP gateway.
SIP negotiates media exchange via Session Description Protocol (SDP), where one side offers a set of capabilities to which the other side answers, thus converging on a set of media characteristics. SIP allows the initial offer to be sent either by the caller in the initial INVITE message (Early Offer) or, if the caller chooses not to, the called party can send the initial offer in the first reliable response (Delayed Offer).
By default, Unified CM SIP trunks send the INVITE without an initial offer (Delayed Offer). Unified CM has two configurable options to enable a SIP trunk to send the offer in the INVITE (Early Offer):
Media Termination Point Required
Checking this option on the SIP trunk assigns an MTP for every outbound call. This statically assigned MTP supports only the G.711 codec or the G.729 codec, thus limiting media to voice calls only.
Early Offer support for voice and video calls (insert MTP if needed)
Checking this option on the SIP Profile associated with the SIP Trunk inserts an MTP only if the calling device cannot provide Unified CM with the media characteristics required to create the Early Offer (for example, where an inbound call to Unified CM is received on a Delayed Offer SIP trunk or a Slow Start H.323 trunk). This option is available only with Unified CM 8.5 and later releases.
In general, Cisco recommends
Early Offer support for voice and video calls (insert MTP if needed)
because this configuration option reduces MTP usage. Calls from older SCCP phones registered to Unified CM over SIP Early Offer trunks use an MTP to create the Offer SDP. These calls support voice, video, and encryption. Inbound calls to Unified CM from SIP Delayed Offer trunks or H.323 Slow Start trunks that are extended over a SIP Early Offer trunk use an MTP to create the Offer SDP. However, these calls support audio only in the initial call setup, but they can be escalated to support video mid-call if the called or calling device invokes it.
Also note that MTP resources are not required for incoming INVITE messages, whether or not they contain an initial offer.
Whether or not an MTP will be allocated by Unified CM depends on the capabilities of the communicating endpoints and the configuration on the intermediary device, if any. For example, the SIP trunk may be configured to handle DTMF exchange in one of several ways: a SIP trunk can carry DTMF using KPML or it can instruct the communicating endpoints to use NTE.
Note As described in this section, SIP Early Offer can also be enabled by checking the Media Termination Point Required option on the SIP trunk. However, this option increases MTP usage because an MTP is assigned for every outbound call rather than on an as-needed basis.
SIP Trunk MTP Requirements
By default, the SIP trunk parameter
Media Termination Point Required
and the SIP Profile parameter
Early Offer support for voice and video calls (insert MTP if needed)
are not selected.
Use the following steps to determine whether MTP resources are required for your SIP trunks.
1. Is the far-end SIP device defined by this SIP trunk capable of accepting an inbound call without a SIP Early Offer?
If not, then on the SIP Profile associated with this trunk, check the box to enable
Early Offer support for voice and video calls (insert MTP if needed)
. For outbound SIP trunk calls, an MTP will be inserted only if the calling device cannot provide Unified CM with the media characteristics required to create the Early Offer, or if DTMF conversion is needed.
If yes, then do not check the
Early Offer support for voice and video calls (insert MTP if needed)
box, and use Step
to determine whether an MTP is inserted dynamically for DTMF conversion. Note that DTMF conversion can be performed by the MTP regardless of the codec in use.
Note The option for Early Offer support for voice and video calls (insert MTP if needed) is available only with Unified CM 8.5 and later releases.
2. Select a Trunk DTMF Signaling Method, which controls the behavior of DTMF selection on that trunk. Available MTPs will be allocated based on the requirements for matching DTMF methods for all calls.
a. DTMF Signaling Method: No Preference
In this mode, Unified CM attempts to minimize the usage of MTP by selecting the most appropriate DTMF signaling method.
If both endpoints support NTE, then no MTP is required.
If both devices support any out-of-band DTMF mechanism, then Unified CM will use KPML or Unsolicited Notify over the SIP trunk. For example, this is the case if a Cisco Unified IP Phone 7936 using SCCP (which supports DTMF using only SCCP messaging) communicates with a Cisco Unified IP Phone 7970 using SIP (which supports DTMF using NTE and KPML) over a SIP trunk configured as described above. The only case where MTP is required is when one of the endpoints supports out-of-band only and the other supports NTE only (for example, an SCCP Cisco Unified IP Phone 7936 communicating with a SIP Cisco Unified IP Phone 7970).
b. DTMF Signaling Method: RFC 2833
By placing a restriction on the DTMF signaling method across the trunk, Unified CM is forced to allocate an MTP if any one or both the endpoints do not support NTE. In this configuration, the only time an MTP will not be allocated is when both endpoints support NTE.
c. DTMF Signaling Method: OOB and RFC 2833
In this mode, the SIP trunk signals both KPML (or Unsolicited Notify) and NTE-based DTMF across the trunk, and it is the most intensive MTP usage mode. The only cases where MTP resources will not be required is when both endpoints support both NTE and any OOB DTMF method (KPML or SCCP).
Note Cisco Unified IP Phones play DTMF to the end user when DTMF is received via SCCP, but they do not play tones received by NTE. However, there is no requirement to send DTMF to another end user. It is necessary only to consider the endpoints that originate calls combined with endpoints that might need DTMF, such as PSTN gateways, application servers, and so forth.
DTMF Relay on SIP Gateways and Cisco Unified Border Element
Cisco SIP Gateways support KPML, NTE, or Unsolicited Notify as the DTMF mechanism, depending on the configuration. Because there may be a mix of endpoints in the system, multiple methods may be configured on the gateway simultaneously in order to minimize MTP requirements.
On Cisco SIP Gateways, configure both
as DTMF relay methods under SIP dial peers. This configuration will enable DTMF exchange with all types of endpoints, including those that support only NTE and those that support only OOB methods, without the need for MTP resources. With this configuration, the gateway will negotiate both NTE and KPML with Unified CM. If NTE is not supported by the Unified CM endpoint, then KPML will be used for DTMF exchange. If both methods are negotiated successfully, the gateway will rely on NTE to receive digits and will not subscribe to KPML.
Cisco SIP gateways also have the ability to use proprietary Unsolicited Notify (UN) method for DTMF. The UN method sends a SIP Notify message with a body that contains text describing the DTMF tone. This method is also supported on Unified CM and may be used if
is not available. Configure
as the DTMF relay method. Note that this method is Cisco proprietary.
SIP gateways that support only NTE require MTP resources to be allocated when communicating with endpoints that do not support NTE.
H.323 Trunks and Gateways
For the H.323 gateways and trunks there are three reasons for invoking an MTP:
H.323 Supplementary Services
MTPs can be used to extend supplementary services to H.323 endpoints that do not support the H.323v2 OpenLogicalChannel and CloseLogicalChannel request features of the Empty Capabilities Set (ECS). This requirement occurs infrequently. All Cisco H.323 endpoints support ECS, and most third-party endpoints have support as well. When needed, an MTP is allocated and connected into a call on behalf of an H.323 endpoint. When an MTP is required on an H.323 call and none is available, the call will proceed but will not be able to invoke supplementary services.
H.323 Outbound Fast Connect
H.323 defines a procedure called Fast Connect, which reduces the number of packets exchanged during a call setup, thereby reducing the amount of time for media to be established. This procedure uses Fast Start elements for control channel signaling, and it is useful when two devices that are utilizing H.323 have high network latency between them because the time to establish media depends on that latency. Unified CM distinguishes between inbound and outbound Fast Start based on the direction of the call setup, and the distinction is important because the MTP requirements are not equal. For inbound Fast Start, no MTP is required. Outbound calls on an H.323 trunk do require an MTP when Fast Start is enabled. Frequently, it is only inbound calls that are problematic, and it is possible to use inbound Fast Start to solve the issue without also enabling outbound Fast Start.
An H.323 trunk supports the signaling of DTMF by means of H.245 out-of-band methods. H.323 intercluster trunks also support DTMF by means of NTE. There are no DTMF configuration options for H.323 trunks; Unified CM dynamically chooses the DTMF transport method.
The following scenarios can occur when two endpoints on different clusters are connected with an H.323 trunk:
When both endpoints are SIP, then NTE is used. No MTP is required for DTMF.
When one endpoint is SIP and supports both KPML and NTE, but the other endpoint is not SIP, then DTMF is sent as KPML from the SIP endpoint to Unified CM, and H.245 is used on the trunk. No MTP is required for DTMF.
If one endpoint is SIP and supports only NTE but the other is not SIP, then H.245 is used on the trunk. An available MTP is allocated for the call. The MTP will be allocated on the Unified CM cluster where the SIP endpoint is located.
For example: A Cisco Unified IP Phone 7970 using SIP to communicate with a Cisco Unified IP Phone 7970 running SCCP, will use NTE when connected via a SIP trunk but will use OOB methods when communicating over an H.323 trunk (with the trunk using the H.245 method).
When a call is inbound from one H.323 trunk and is routed to another H.323 trunk, NTE will be used for DTMF when both endpoints are SIP. H.245 will be used if either endpoint is not SIP. An MTP will be allocated if one side is a SIP endpoint that supports only NTE and the other side is non-SIP.
DTMF Relay on H.323 Gateways and Cisco Unified Border Element
H.323 gateways support DTMF relay via H.245 Alphanumeric, H.245 Signal, NTE, and audio in the media stream. The NTE option must not be used because it is not supported on Unified CM for H.323 gateways at this time. The preferred option is H.245 Signal. MTPs are required for establishing calls to an H.323 gateway if the other endpoint does not have signaling capability in common with Unified CM. For example, a Cisco Unified IP Phone 7960 running the SIP stack supports only NTEs, so an MTP is needed with an H.323 gateway.
CTI Route Points
A CTI Route Point uses CTI events to communicate with CTI applications. For DTMF purposes, the CTI Route Point can be considered as an endpoint that supports all OOB methods and does not support RFC 2833. For such endpoints, the only instance where an MTP will be required for DTMF conversion would be when it is communicating with another endpoint that supports only RFC 2833.
CTI Route Points that have first-party control of a phone call will participate in the media stream of the call and require an MTP to be inserted. When the CTI has third-party control of a call so that the media passes through a device that is controlled by the CTI, then the requirement for an MTP is dependent on the capabilities of the controlled device.
Example 7-1 Call Flow that Requires an MTP for NTE Conversion
Assume the example system has CTI route points with first-party control (the CTI port terminates the media), which integrate to a system that uses DTMF to navigate an IVR menu. If all phones in the system are running SCCP, then no MTP is required. In this case Unified CM controls the CTI port and receives DTMF from the IP phones via SCCP. Unified CM provides DTMF conversion.
However, if there are phones running a SIP stack (that support only NTE and not KPML), an MTP is required. NTEs are part of the media stream; therefore Unified CM does not receive them. An MTP is invoked into the media stream and has one call leg that uses SCCP, and the second call leg uses NTEs. The MTP is under SCCP control by Unified CM and performs the NTE-to-SCCP conversion. Note that the newer phones that do support KPML will not need an MTP.
MTP Usage with a Conference Bridge
MTPs are utilized in a conference call when one or more participant devices in the conference use RFC 2833. When the conference feature is invoked, Unified CM allocates MTP resources for every conference participant device in the call that supports only RFC 2833. This is regardless of the DTMF capabilities of the conference bridge used.
The following types of devices are available for use as an MTP:
Software MTP (Cisco IP Voice Media Streaming Application)
A software MTP is a device that is implemented by enabling the Cisco IP Voice Media Streaming Application on a Unified CM server. When the installed application is configured as an MTP application, it registers with a Unified CM node and informs Unified CM of how many MTP resources it supports. A software MTP device supports only G.711 streams. The IP Voice Media Streaming Application is a resource that may also be used for several functions, and the design guidance must consider all functions together (see Cisco IP Voice Media Streaming Application).
Software MTP (Based on Cisco IOS)
The capability to provide a software-based MTP on the router is available beginning with Cisco IOS Release 12.3(11)T for the Cisco 3800 Series Routers; Release 15.0(1)M for the Cisco 2900 Series and 3900 Series Routers; Release IOS-XE for ASR1002, 1004, and 1006 Routers; Release IOS-XE 3.2 for ASR1001 Routers; and Release 12.3(8)T4 for other router models.
This MTP allows configuration of any of the following codecs, but only one may be configured at a given time: G.711 mu-law and a-law, G.729a, G.729, G.729ab, G.729b, and passthrough. Some of these are not pertinent to a Unified CM implementation.
Router configurations permit up to 1,000 individual streams, which support 500 transcoded sessions. This number of G.711 streams generates 10 Mbytes of traffic. The Cisco ISR G2s and ASR routers can support significantly higher numbers than this.
Hardware MTP (PVDM2, Cisco NM-HDV2 and NM-HD-1V/2V/2VE)
This hardware uses the PVDM-2 modules for providing DSPs.
Each DSP can provide 16 G.711 mu-law or a-law, 8 G.729a or G.722, or 6 G.729 or G.729b MTP sessions.
Hardware MTP (Cisco 2900 and 3900 Series Routers with PVDM3)
These routers use the PVDM3 DSPs natively on the motherboards or PVDM2 with an adaptor on the motherboard or on service modules.
The capacity of each of the DSP type varies from 16 G.711 a-law or mu-law sessions for the PVDM3-16 to 256 G.711 sessions for the PVDM3-256.
Note You cannot configure G.729 or G.729b codecs when configuring hardware MTP resources in Cisco IOS. However, Unified CM can use hardware transcoding resources as MTPs if all other MTP resources are exhausted or otherwise unavailable.
An annunciator is a software function of the Cisco IP Voice Media Streaming Application that provides the ability to stream spoken messages or various call progress tones from the system to a user. It uses SCCP messages to establish RTP streams, and it can send multiple one-way RTP streams to devices such as Cisco IP phones or gateways. The device must be capable of SCCP to utilize this feature. SIP phones and devices are still able to receive all the various messages provided by the annunciator. For SIP devices, all these messages and tones are downloaded (pushed) to the device at registration so that they can be invoked as needed by SIP signaling messages from Unified CM.
Tones and announcements are predefined by the system. The announcements support localization and may also be customized by replacing the appropriate .wav file. The annunciator is capable of supporting G.711 a-law and mu-law, G.729, and Wideband codecs without any transcoding resources.
The following features require an annunciator resource:
Cisco Multilevel Precedence Preemption (MLPP)
This feature has streaming messages that it plays in response to the following call failure conditions.
– Unable to preempt due to an existing higher-precedence call.
– A precedence access limitation was reached.
– The attempted precedence level was unauthorized.
– The called number is not equipped for preemption or call waiting.
Integration via SIP trunk
SIP endpoints have the ability to generate and send tones in-band in the RTP stream. Because SCCP devices do not have this ability, an annunciator is used in conjunction with an MTP to generate or accept DTMF tones when integrating with a SIP endpoint. The following types of tones are supported:
– Call progress tones (busy, alerting, and ringback)
– DTMF tones
Cisco IOS gateways and intercluster trunks
These devices require support for call progress tone (ringback tone).
During the following call failure conditions, the system plays a streaming message to the end user:
– A dialed number that the system cannot recognize
– A call that is not routed due to a service disruption
– A number that is busy and not configured for preemption or call waiting
During a conference call, the system plays a barge-in tone to announce that a participant has joined or left the bridge.
An annunciator is automatically created in the system when the Cisco IP Voice Media Streaming Application is activated on a server. If the Media Streaming Application is deactivated, then the annunciator is also deleted. A single annunciator instance can service the entire Unified CM cluster if it meets the performance requirements (see Annunciator Performance); otherwise, you must configure additional annunciators for the cluster. Additional annunciators can be added by activating the Cisco IP Voice Media Streaming Application on other servers within the cluster.
The annunciator registers with a single Unified CM at a time, as defined by its device pool. It will automatically fail over to a secondary Unified CM if a secondary is configured for the device pool. Any announcement that is playing at the time of an outage will not be maintained.
An annunciator is considered a media device, and it can be included in media resource groups (MRGs) to control which annunciator is selected for use by phones and gateways.
By default, the annunciator is configured to support 48 simultaneous streams, which is the maximum recommended for an annunciator running on the same server (co-resident) with the Unified CM service. If the server has only 10 Mbps connectivity, lower the setting to 24 simultaneous streams.
A standalone server without the Cisco CallManager Service can support up to 255 simultaneous announcement streams, and a high-performance server with dual CPUs and a high-performance disk system can support up to 400 streams. You can add multiple standalone servers to support the required number of streams.
Music on Hold
The Music on Hold (MoH) feature requires that each MoH server must be part of a Unified CM cluster and participate in the data replication schema. Specifically, the MoH server must share the following information with the Unified CM cluster through the database replication process:
Audio sources - The number and identity of all configured MoH audio sources
Multicast or unicast - The transport nature (multicast or unicast) configured for each of these sources
Multicast address - The multicast base IP address of those sources configured to stream as multicast
To configure a MoH server, enable the Cisco IP Voice Media Streaming Application Service on one or more Unified CM nodes. An MoH server can be deployed along with Unified CM on the same server or in standalone mode.
Unicast and Multicast MoH
Unified CM supports unicast and multicast MoH transport mechanisms.
A unicast MoH stream is a point-to-point, one-way audio Real-Time Transport Protocol (RTP) stream from the MoH server to the endpoint requesting MoH. It uses a separate source stream for each user or connection. Thus, if twenty devices are on hold, then twenty streams are generated over the network between the server and these endpoint devices. Unicast MoH can be extremely useful in those networks where multicast is not enabled or where devices are not capable of multicast, thereby still allowing an administrator to take advantage of the MoH feature. However, these additional MoH streams can potentially have a negative effect on network throughput and bandwidth.
A multicast MoH stream is a point-to-multipoint, one-way audio RTP stream between the MoH server and the multicast group IP address. The endpoints requesting an MoH audio stream can join the multicast group as needed. This mode of MoH conserves system resources and bandwidth because it enables multiple users to use the same audio source stream to provide music on hold. For this reason, multicast is an extremely attractive transport mechanism for the deployment of a service such as MoH because it greatly reduces the CPU impact on the source device and also greatly reduces the bandwidth consumption for delivery over common paths. However, multicast MoH can be problematic in situations where a network is not enabled for multicast or where the endpoint devices are not capable of handling multicast.
There are distinct differences between unicast and multicast MoH in terms of call flow behavior. A unicast MoH call flow is initiated by a message from Unified CM to the MoH server. This message tells the MoH server to send an audio stream to the holdee device's IP address. On the other hand, a multicast MoH call flow is initiated by a message from Unified CM to the holdee device. This message instructs the endpoint device to join the multicast group address of the configured multicast MoH audio stream.
For a detailed look at MoH call flows, see the section on MoH Call Flows.
Supported Unicast and Multicast Gateways
The following gateways support both unicast and multicast MoH:
Cisco 2900 Series and Cisco 3900/3900E Series ISR G2 Routers with Cisco IOS 15.0.1M or later release
Cisco 2800 Series and 3800 Series Routers with MGCP or H.323 and Cisco IOS 12.3.14T or later release
Cisco 2800 Series and 3800 Series Routers with SIP and Cisco IOS 12.4(24)T or later release
Cisco VG224 Analog Voice Gateways with MGCP and Cisco IOS 12.3.14T or later release
Cisco VG204 and VG202 Analog Voice Gateways with MGCP or SCCP and Cisco IOS 12.4(22)T or later release
Cisco VG248 Analog Phone Gateways
Cisco ASR 1000 Series Aggregation Services Routers
Note Cisco 2800 Series, 3800 Series, and VG248 gateways are End of Sale (EoS). There are other legacy gateways that also support unicast and multicast MoH.
Note The Cisco Unified Border Element on Cisco ASR 1000 Series Aggregation Services Routers might not support one-way streaming of music or announcements by the Cisco Unified Communications Manager Music on Hold (MoH) feature. For more information, refer to the release notes for your version of Cisco Unified Communications Manager, available at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_release_notes_list.html.
MoH Selection Process
This section describes the MoH selection process as implemented in Unified CM.
The basic operation of MoH in a Cisco Unified Communications environment consists of a holder and a holdee. The
is the endpoint user or network application placing a call on hold, and the
is the endpoint user or device placed on hold.
The MoH stream that an endpoint receives is determined by a combination of the User Hold MoH Audio Source of the device placing the endpoint on hold (holder) and the configured media resource group list (MRGL) of the endpoint placed on hold (holdee). The User Hold MoH Audio Source configured for the holder determines the audio file that will be streamed when the holder puts a call on hold, and the holdee's configured MRGL indicates the resource or server from which the holdee will receive the MoH stream.
As illustrated by the example in Figure 7-3, if phones A and B are on a call and phone B (holder) places phone A (holdee) on hold, phone A will hear the MoH audio source configured for phone B (Audio-source2). However, phone A will receive this MoH audio stream from the MRGL (resource or server) configured for phone A (MRGL A).
Figure 7-3 User Hold Audio Source and Media Resource Group List (MRGL)
Because the configured MRGL determines the server from which a unicast-only device will receive the MoH stream, you must configure unicast-only devices with an MRGL that points to a unicast MoH resource or media resource group (MRG). Likewise, a device capable of multicast should be configured with an MRGL that points to a multicast MRG containing a MoH server configured for multicast.
User and Network Hold
User hold includes the following types:
User on hold at an IP phone or other endpoint device
User on hold at the PSTN, where MoH is streamed to the gateway
Figure 7-4 shows these two types of call flows. If phone A is in a call with phone B and phone A (holder) pushes the Hold softkey, then a music stream is sent from the MoH server to phone B (holdee). The music stream can be sent to holdees within the IP network or holdees on the PSTN, as is the case if phone A places phone C on hold. In the case of phone C, the MoH stream is sent to the voice gateway interface and converted to the appropriate format for the PSTN phone. When phone A presses the Resume softkey, the holdee (phone B or C) disconnects from the music stream and reconnects to phone A.
Figure 7-4 Basic User Hold Example
Network hold can occur in following scenarios:
Figure 7-5 illustrates an example of network hold during a call transfer. The call flow involves the following steps:
1. Phone A receives a call from PSTN phone C.
2. Phone A answers the call and then transfers it to phone B. During the transfer process, phone C is put on network hold.
3. Phone C receives an MoH stream from the MoH server via the gateway. After phone A completes the transfer action, phone C disconnects from the music stream and gets redirected to phone B.
This process is the same for other network hold operations such as call park and conference setup.
Figure 7-5 Basic Network Hold Example for Call Transfer
A Unified CM MoH server can generate a MoH stream from two types of sources, audio file and fixed source, either of which can be transmitted as unicast or multicast. You can configure a maximum of 51 MoH audio sources per Unified CM cluster, of which up to 50 can be audio files but only one can be a fixed source.
Audio files (.wav format) can be uploaded to Unified CM, which then automatically generates MoH audio files for the specified codecs. Unified CM supports G711 (a-law and mu-law), G.729 Annex A, and Wideband codecs for MoH streams.
Note Before configuring a MoH audio source, you must upload the .wav formatted audio source file to every MoH server within the cluster using the upload file function in the Unified CM Administration interface. Cisco recommends that you first upload the audio source file onto each MoH server in the cluster, then upload it onto the publisher (even if not an MoH server), and finally assign an MoH Audio Stream Number and configure the MoH audio source in the Unified CM Administration interface on the publisher.
If recorded or live audio is needed, MoH can be generated from a fixed source connected to the audio input of the local sound card. The Cisco MoH USB audio sound card (MOH-USB-AUDIO=) must be used for connecting a fixed or live audio source to the MoH server. This USB sound card is compatible with all Cisco MCS platforms that support Cisco Unified CM.
This mechanism enables you to use radios, CD players, or any other compatible sound source to stream MoH. The stream from the fixed audio source is transcoded in real-time to support the codec that was configured through Unified CM Administration. The fixed audio source can be transcoded into G.711 (A-law or mu-law), G.729 Annex A, and Wideband, and it is the only audio source that is transcoded in real-time.
Note Prior to using a fixed audio source to transmit music on hold, you should consider the legalities and the ramifications of re-broadcasting copyrighted audio materials. Consult your legal department for potential issues.
To determine which User and Network Audio Source configuration setting to apply in a particular case, Unified CM interprets these settings for the
device in the following priority order:
1. Directory or line setting (Devices with no line definition, such as gateways, do not have this level.)
2. Device setting
3. Common Device Configuration setting
4. Cluster-wide default setting
Unified CM also interprets the MRGL configuration settings of the
device in the following priority order:
1. Device setting
2. Device pool setting
3. System default MoH resources
Note that system default MoH resources are resources that are not assigned to any MRG and they are always unicast.
MoH Call Flows
The following sections provide detailed illustrations and explanations of unicast and multicast MoH call flows for both SCCP and SIP endpoints.
SCCP Call Flows
This section describes the multicast and unicast call flows for music on hold with Skinny Client Control Protocol (SCCP) endpoints.
SCCP Multicast Call Flow
Figure 7-6 illustrates a typical SCCP multicast call flow. As shown in the diagram, when the Hold softkey is pressed at phone A, Unified CM instructs both phone A and phone B to Close Receive Channel and Stop Media Transmission. This action effectively stops the RTP two-way audio stream. Next, Unified CM tells phone B (the holdee) to Start Multicast Media Reception from multicast group address 126.96.36.199. The phone then issues an Internet Group Management Protocol (IGMP) V2 Membership Report message indicating that it is joining this group.
Figure 7-6 Detailed SCCP Multicast MoH Call Flow
Meanwhile, the MoH server has been sourcing RTP audio to this multicast group address and, upon joining the multicast group, phone B begins receiving the MoH stream. Once phone A presses the Resume softkey, Unified CM instructs phone B to Stop Multicast Media Reception. Phone B then sends an IGMP V2 Leave Group message to 188.8.131.52 to indicate that the multicast stream is no longer needed. This effectively ends the MoH session. Next, Unified CM sends a series of Open Receive Channel messages to phones A and B, just as would be sent at the beginning of a phone call between the two phones. Soon afterwards, Unified CM instructs both phones to Start Media Transmission to each other’s IP addresses. The phones are once again connected by means of an RTP two-way audio stream.
Note The call flow diagrams in Figure 7-6 and Figure 7-7 assume that an initial call exists between phones A and B, with a two-way RTP audio stream. These diagrams are representative of call flows and therefore include only the pertinent traffic required for proper MoH operation. Thus, keep-alives, acknowledgements, and other miscellaneous traffic have been eliminated to better illustrate the interaction. The initial event in each diagram is the Hold softkey action performed by phone A.
SCCP Unicast Call Flow
Figure 7-7 depicts an SCCP unicast MoH call flow. In this call flow diagram, when the Hold softkey is pressed at phone A, Unified CM instructs both phone A and phone B to Close Receive Channel and Stop Media Transmission. This action effectively stops the RTP two-way audio stream. Up to this point, unicast and multicast MoH call flows behave exactly the same way.
Figure 7-7 Detailed SCCP Unicast MoH Call Flow
Next, Unified CM tells phone B (the holdee) to Open Receive Channel. (This is quite different from the multicast case, where Unified CM tells the holdee to Start Multicast Media Reception.) Then Unified CM tells the MoH server to Start Media Transmission to the IP address of phone B. (This too is quite different behavior from the multicast MoH call flow, where the phone is prompted to join a multicast group address.) At this point, the MoH server is sending a one-way unicast RTP music stream to phone B. When phone A presses the Resume softkey, Unified CM instructs the MoH server to Stop Media Transmission and instructs phone B to Close Receive Channel, effectively ending the MoH session. As with the multicast scenario, Unified CM sends a series of Open Receive Channel messages and Start Media Transmissions messages to phones A and B with each other’s IP addresses. The phones are once again connected by means of an RTP two-way audio stream.
SIP Call Flows
This section describes the multicast and unicast call flows for music on hold with Session Initiation Protocol (SIP) endpoints.
SIP Multicast Call Flow
Figure 7-8 illustrates a typical SIP multicast call flow. As shown in the diagram, when the Hold softkey is pressed at phone A, phone A sends a SIP INVITE with a Session Description Protocol (SDP) connection information indication of phone A's IP address and a media attribute indication of sendonly. Unified CM then instructs phone A to disconnect the RTP stream by means of a SIP 200 OK Response with an SDP connection information indication of 0.0.0.0 and a media attribute indication of recvonly. Phone B is then told to disconnect the RTP stream by means of a SIP INVITE from Unified CM with an SDP connection information indication of 0.0.0.0 and a media attribute of inactive. After a SIP 200 OK Response is sent back from phone B to Unified CM indicating an SDP media attribute of inactive, Unified CM then sends a SIP INVITE to phone B with an SDP connection information indication of the MoH multicast group address (in this case 184.108.40.206) and a media attribute of recvonly.
Figure 7-8 Detailed SIP Multicast MoH Call Flow
Next, phone B in Figure 7-8 issues an IGMP V2 Membership Report message indicating that it is joining this multicast group. In addition, phone B sends a SIP 200 OK Response back to Unified CM indicating an SDP media attribute of recvonly in response to the previous SIP INVITE. Meanwhile, the MoH server has been sourcing RTP audio to this MoH multicast group address and, upon joining the multicast group, phone B begins receiving the one-way MoH stream.
When the user at phone A presses the Resume softkey, phone A sends a SIP INVITE with an SDP connection information indication of phone A's IP address and media attribute indications of phone A's receiving RTP port and sendrecv. Unified CM then instructs phone B to disconnect from the multicast MoH stream by means of a SIP INVITE with an SDP connection information indication of 0.0.0.0 and a media attribute indication of inactive. A SIP 200 OK Response is sent back from phone B to Unified CM, indicating an SDP media attribute of inactive.
Next Unified CM sends a SIP INVITE to phone B, and phone B responds with a SIP 200 OK Response with an SDP connection information indication of phone B's IP address and media attribute indications of phone B's receiving RTP port and sendrecv. Unified CM responds by sending a SIP ACK to phone B with an SDP connection information indication of phone A's IP address and a media attribute of phone A's receiving RTP port number. Likewise, Unified CM forwards a SIP 200 OK Response to phone A's original resuming SIP INVITE, with an SDP connection information indication of phone B's IP address and a media attribute of phone B's receiving RTP port number. Phone B then sends an IGMP V2 Leave Group message to 220.127.116.11 to indicate that the multicast stream is no longer needed. Finally, the RTP two-way audio stream between phones A and B is reestablished.
Note The call flow diagrams in Figure 7-8 and Figure 7-9 assume that an initial call exists between phones A and B, with a two-way RTP audio stream. These diagrams are representative of call flows and therefore include only the pertinent traffic required for proper MoH operation. Thus, keep-alives, some acknowledgements, progression indications, and other miscellaneous traffic have been eliminated to better illustrate the interaction. The initial event in each diagram is the Hold softkey action performed by phone A.
SIP Unicast Call Flow
Figure 7-9 depicts a SIP unicast MoH call flow. As shown in the diagram, when the Hold softkey is pressed at phone A, phone A sends a SIP INVITE with an SDP connection information indication of phone A's IP address and a media attribute indication of sendonly. Unified CM then instructs phone A to disconnect the RTP stream by means of a SIP 200 OK Response with an SDP connection information indication of 0.0.0.0 and a media attribute indication of recvonly. Phone B is then told to disconnect the RTP stream by means of a SIP INVITE from Unified CM, with an SDP connection information indication of 0.0.0.0 and a media attribute of inactive. Next a SIP 200 OK Response is sent back from phone B to Unified CM, indicating an SDP media attribute of inactive. Up to this point, unicast and multicast MoH call flows are exactly the same.
Figure 7-9 Detailed SIP Unicast MoH Call Flow
Unified CM then sends a SIP INVITE to phone B, and phone B responds back with a SIP 200 OK Response indicating SDP connection information with phone B's IP address and media attribute indications of phone B's receiving RTP port number and sendrecv. Unified CM then sends a SCCP StartMediaTransmission message to the MoH server, with phone B's address and receiving RTP port number. This is followed by a SIP ACK from Unified CM to phone B indicating SDP connection information of the Unified CM IP address and a media attribute of sendonly. Meanwhile, the MoH server begins sourcing RTP audio to phone B, and phone B begins receiving the one-way MoH stream.
When the user at phone A presses the Resume softkey, phone A sends a SIP INVITE with an SDP connection information indication of phone A's IP address and media attribute indications of phone A's receiving RTP port and sendrecv. Unified CM then instructs phone B to disconnect from the multicast MoH stream by means of a SIP INVITE with an SDP connection information indication of 0.0.0.0 and a media attribute indication of inactive. A SIP 200 OK Response is sent back from phone B to Unified CM, indicating an SDP media attribute of inactive. Then Unified CM sends an SCCP StopMediaTransmission message to the MoH server, causing the MoH server to stop forwarding the MoH stream to phone B.
Next Unified CM sends a SIP INVITE to phone B, and phone B responds with a SIP 200 OK Response with an SDP connection information indication of phone B's IP address and media attribute indications of phone B's receiving RTP port and sendrecv. Unified CM responds by sending a SIP ACK to phone B, with an SDP connection information indication of phone A's IP address and a media attribute of phone A's receiving RTP port number. Likewise, Unified CM forwards a SIP 200 OK Response to phone A's original resuming SIP INVITE with an SDP connection information indication of phone B's IP address and a media attribute of phone B's receiving RTP port. Finally, the RTP two-way audio stream between phones A and B is reestablished.
Capacity Planning for Media Resources
This section provides information on the capacities of various network modules and chassis that carry DSPs, the capacities of the chassis to carry network modules, and software dependencies of the hardware.
For all Cisco ISR G1 and G2 capacity planning, use the DSP Calculator available at
. For other platforms (such as the Cisco 1700, 2600, and 3700 Series Routers), use the legacy DSP calculator at
The DSP resources for Unified Communications solutions are provided by NM-HD, NM-HDV, and PVDM modules. NM-HD and NM-HDV2 modules are supported on Cisco ISR G1 and G2 Series platforms. Refer to the respective product data sheets for capacity information for these modules.
PVDM modules are available in three models: PVDM-256K, PVDM2, and the newer PVDM3. Each of the models has several modules with different density support. For example, a PVDM-256K-4 and a PVDM2-16 are single DSP modules in their respective category. PVDM2 modules are supported on Cisco ISR G1 and ISR G2 platforms (minimum of Cisco IOS Release 15.0(1)M is required for the Cisco ISR G2 Series). The PVDM3 DSP modules are supported on the Cisco 2900 Series and 3900 Series platforms and require a minimum Cisco IOS Release of 15.0(1) M. PVDM3 modules provide DSP resources for both voice and video. The PVDM3 modules are newer than the PVDM2 and PVDM-256K modules, and the three types are not interchangeable.
Some things to consider when doing capacity planning for hardware-based media resources include the density of the module, the underlying platform (Cisco ISR G1 or G2), and the minimum Cisco IOS version required.
For capacity information on PVDM2 modules, refer to the
High-Density Packet Voice Digital Signal Processor Module for Cisco Unified Communications Solutions
data sheet, available at
For capacity information on PVDM3 modules, refer to the
High-Density Packet Voice Video Digital Signal Processor Module for Cisco Unified Communications Solutions
data sheet, available at
Considerations for Cisco 2900 and 3900 Series Platforms
The following guidelines and considerations apply to the DSP resources hosted by these platforms:
The Cisco 2900 and 3900 Series Routers support only the PVDM3 DSPs in the on-board (motherboard) DSP slots. PVDM2 DSPs may be used in those slots by using an adaptor card. NM-HD and NM-HDV2 cards can be used in Service Module slots with an adaptor card.
PVDM2 and PVDM3 modules cannot be used at the same time on the same motherboard.
DSP sharing can be done only between the same DSP types. For example, if the motherboard is populated with PVDM3 DSPs and the Service Modules are populated with PVDM2 DSPs, then the DSPs in the Service Modules may be shared with each other but DSPs on the motherboard may not be shared with those in the Service Modules.
PVDM3 DSPs support all the functions that the PVDM2 DSPs support except for Cisco Fax Relay.
Unlike the PVDM2, the PVDM3 DSPs have a single software image for all media functions.
Cisco 2800 and 3800 Series Platforms
The following guidelines and considerations apply to the DSP resources hosted by these platforms:
Although the Cisco 2800 and 3800 Series Routers all have two AIM slots, they do not support the AIM-VOICE-30 or AIM-ATM-VOICE-30 cards because PVDM2 modules that are installed on the motherboard provide that functionality.
You can install the NM-HDV2, NM-HD-
, and NM-HDV modules in the Cisco IOS platforms as indicated in the product data sheets.
All three families of modules may be installed in a single chassis. However, the conferencing and transcoding features cannot be used simultaneously on both the NM-HDV family and either of the other two families (NM-HD-
or NM-HDV2). In addition, the NM-HDV (TI-549), NM-HD-
, and NM-HDV2 (TI-5510) cannot be used simultaneously for conferencing and transcoding within a single chassis.
You can mix NM-HDV and NM-HDV-FARM modules in the same chassis, but not all chassis can be completely populated by these modules.
Capacity Planning for Music on Hold
It is important to be aware of the hardware capacity for MoH resources and to consider the implications of multicast and unicast MoH in relation to this capacity when doing capacity planning for MoH resources. The capacity of the MoH server depends on several factors such as deployment model (co-resident or standalone), underlying server platform, and so forth.
Co-resident and Standalone MoH
The MoH feature requires the use of a server that is part of a Unified CM cluster. You can configure the MoH server in either of the following ways:
refers to two or more services or applications running on the same server. In a co-resident deployment, the MoH feature runs on any server (either publisher or subscriber) in the cluster that is also running the Unified CM software.
A standalone deployment, places the MoH feature on a dedicated media resource server node within the Unified CM cluster. This server acts as neither a publisher or a subscriber. That is, the Cisco IP Voice Media Streaming Application service is the only service enabled on the server. The only function of this dedicated server is to send MoH streams to devices within the network.
Server Platform Limits
Starting with Cisco Unified Communications Manager Release 9.0, a maximum of 1,000 MoH streams is supported across MCS 7835 and MCS 7845 servers or OVA equivalent platforms for both standalone and co-resident deployments. For other platforms, Unified CM can support half that amount or less, depending upon what other services are active on the server. Ensure that network call volumes do not exceed these limits because, once MoH sessions have reached these limits, additional load could result in poor MoH quality, erratic MoH operation, or even loss of MoH functionality. Note that you can configure a maximum of 51 unique audio sources per Unified CM cluster.
For more information on supported server platforms, refer to
Supported Servers for Releases of Cisco Unified Communications Manager (Including Business Edition 3000/5000/6000 and Session Manager Edition) and Cisco Intercompany Media Engine
, available at
The following two MoH Server Configuration parameters affect MoH server capacity:
Maximum Half Duplex Streams
This parameter determines the number of devices that can be placed on unicast MoH. By default this value is set to 250.
The Maximum Half Duplex Streams parameter should be set to the value derived from the following formula:
(Server and deployment capacity) – ((Number of multicast MoH sources) ∗ (Number of MoH codecs enabled))
MCS-7835 standalone MoH server (or OVA equivalent)
Multicast MoH audio sources
MoH codecs enabled (G.711 mu-law and G.729)
Maximum half-duplex streams
Therefore, in this example, the Maximum Half Duplex Streams parameter would be configured with a value of no more than 976.
Maximum Multicast Connections
This parameter determines the number of devices that can be placed on multicast MoH.
The Maximum Multicast Connections parameter should be set to a number that ensures that all devices can be placed on multicast MoH if necessary. Although the MoH server can generate only a finite number of multicast streams, a large number of held devices can join each multicast stream. This parameter should be set to a number that is greater than or equal to the number of devices that might be placed on multicast MoH at any given time. Typically multicast traffic is accounted for based on the number of streams being generated; however, Unified CM maintains a count of the actual number of devices placed on multicast MoH or joined to each multicast MoH stream. Although this method is different than the way multicast traffic is normally tracked, it is important to configure this parameter appropriately.
Note Because you can configure only 51 unique audio sources per Unified CM cluster and because there are only four possible codecs for MoH streams, the maximum number of multicast streams per MoH server is 204.
Failure to configure these parameters properly could lead to under-utilization of MoH server resources or failure of the server to handle the network load. For details on how to configure the service parameters, refer to the
Cisco Unified Communications Manager Administration Guide
, available at
Note The maximum limit of 1,000 sessions per MoH server applies to unicast, multicast, or simultaneous unicast and multicast sessions. The limit represents the recommended maximum number of sessions a platform can support, irrespective of the transport mechanism.
When provisioning for co-resident or standalone MoH server configurations, network administrators should consider the type of transport mechanism used for the MoH audio streams. If using unicast MoH, each device on hold requires a separate MoH stream. However, if using multicast MoH and only a single audio source, then only a single MoH stream is required for each configured codec type, no matter how many devices of that type are on hold.
For example, given a cluster with 30,000 phones and a 2% hold rate (only 2% of all endpoint devices are on hold at any given time), 600 MoH streams or sessions would be required. Given a unicast-only MoH environment, one co-resident (or standalone) MoH server running on an MCS 7845 (or OVA equivalent) would be required to handle this load.
By comparison, a multicast-only MoH environment with 36 unique MoH audio streams, for example, would require one co-resident MoH server (MCS 7816, 7825, or 7878). These 36 unique multicast streams could be provisioned in any one of the following ways:
36 unique audio sources streamed using a single codec
18 unique audio sources streamed using only 2 codecs
12 unique audio sources streamed using only 3 codecs
9 unique audio source streamed using all 4 codecs
In the preceding examples, the 2% hold rate is based on 30,000 phones and does not take into account gateways or other endpoint devices in the network that are also capable of being placed on hold. You should consider these other devices when calculating a hold rate because they could potentially be placed on hold just as the phones can.
The preceding calculations also do not provide for MoH server redundancy. If an MoH server fails or if more than 2% of the users go on hold at the same time, there are no other MoH resources in this scenario to handle the overflow or additional load. Your MoH resource calculations should include enough extra capacity to provide for redundancy. Additional MoH servers can be provisioned for redundancy or high availability as explained in the section on High Availability for Media Resources.
Design Considerations for Media Resources
This section discusses specific considerations for deploying media resources for use with the various Unified CM deployment models. It also highlights the configuration considerations and best practices to help you design a robust solution for media resource allocation in your Unified CM implementation.
This section examines where and when the MTP and transcoding resources are used within the following three enterprise IP Telephony deployment models:
In a single-site deployment, there is no need for transcoding because there are no low-speed links to justify the use of a low bit-rate (LBR) codec. Some MTP resources might be required in the presence of a significant number of devices that are not compliant with H.323v2, such as older versions of Microsoft NetMeeting or certain video devices. MTP resources may be required for DTMF conversion if SIP endpoints are present (see Named Telephony Events (RFC 2833).)
Multisite Deployments with Centralized Call Processing
In a centralized call processing deployment, the Unified CM cluster and the applications (such as voice mail and IVR) are located at the central site, while several remote sites are connected through an IP WAN. The remote sites rely on the centralized Unified CMs to handle their call processing.
Because WAN bandwidth is typically limited, calls are configured to use a low bit-rate codec such as G.729 when traversing the WAN. (See Figure 7-10.)
Voice compression between IP phones is easily configured through the use of
in Unified CM. A region defines the type of compression (for example, G.711 or G.729) used by the devices in that region, and a location specifies the total amount of bandwidth available for calls to and from devices at that location.
Figure 7-10 Transcoding for the WAN with Centralized Call Processing
Unified CM uses media resource groups (MRGs) to enable sharing of MTP and transcoding resources among the Unified CM servers within a cluster. In addition, when using an LBR codec (for example, G.729a) for calls that traverse different regions, the transcoding resources are used only if one (or both) of the endpoints is unable to use the LBR codec.
In Figure 7-10, Unified CM knows that a transcoder is required and allocates one based on the MRGL and/or MRG of the device that is using the higher-bandwidth codec. In this case it is the VM/UM server that determines which transcoder device is used. This behavior of Unified CM is based on the assumption that the transcoder resources are actually located close to the higher-bandwidth device. If this system was designed so that the transcoder for the VM/UM server was located at the remote site, then G.711 would be sent across the WAN, which would defeat the intended design. As a result, if there are multiple sites with G.711-only devices, then each of these sites would need transcoder resources when an LBR is run on the WAN.
The placement of other resources is also important. For example, if a conference occurs with three phones at a remote site and the conference resource is located in the central (call processing) site, then three media streams are carried over the WAN. If the conference resource were local, then the calls would not traverse the WAN. It is necessary to consider this factor when designing the bandwidth and call admission control for your WAN.
Multisite Deployments with Distributed Call Processing
In distributed call processing deployments, several sites are connected through an IP WAN. Each site contains a Unified CM cluster that can, in turn, follow the single-site model or the centralized call processing model. A gatekeeper may be used for call admission control between sites.
Because WAN bandwidth is typically limited, calls between sites may be configured to use an LBR codec (such as G.729a) when traversing the WAN. H.323v2 intercluster trunks are used to connect Unified CM clusters. Unified CM also supports compressed voice call connections through the MTP service if a hardware MTP is used. (See Figure 7-11.)
A distributed call processing deployment might need transcoding and MTP services in the following situations:
With current versions of Cisco applications, it is possible and recommended to avoid the use of transcoding resources. There might be specific instances where G.711 on a specific device cannot be avoided.
Some endpoints (for example, video endpoints) do not support the H.323v2 features.
Figure 7-11 Intercluster Call Flow with Transcoding
Unified CM uses media resource groups (MRGs) to enable sharing of MTP and transcoding resources among the Unified CM servers within a cluster. In addition, for calls across intercluster trunks, MTP and transcoding resources are used only when needed, thus eliminating the need to configure the MTP service for applications that do not support LBR codecs.
The following characteristics apply to distributed call processing deployments:
Only the intercluster calls that require transcoding will use the MTP service. For example, if both endpoints of a call are capable of using a G.729 codec, no transcoding resources will be used.
Sharing MTP resources among servers within a cluster provides more efficient resource utilization.
Media Functions and Voice Quality
Any process that manipulates media can degrade the quality of the media. For example, encoding a voice stream for transmission across any network (IP or TDM) and decoding it at the other end will result in a loss of information, and the resulting voice stream will not be an exact reproduction of the original. If there are media traversal paths through the network that involve multiple encoding and decoding steps of the same voice stream, then each successive encoding/decoding operation will further degrade the voice quality. In general, such paths should be avoided. This is especially true for low-bandwidth codecs (LBC) such as G.729.
If such paths cannot be avoided, voice quality can generally be improved by using a higher bandwidth, low-compression codec, such as the G.711 or G.722 codecs, which are recommended wherever such paths are anticipated. Use of lower bandwidth, higher compression codecs in such scenarios is not recommended.
Music on Hold Design Considerations
This section highlights some MoH configuration considerations and best practice to help you design a robust MoH solution.
If you need multiple codecs for MoH deployment, configure them in the IP Voice Media Streaming Application service parameter
Supported MoH Codecs
under the Clusterwide Unified CM Service Parameters Configuration. From the Supported MoH Codecs list under the Clusterwide Parameters, select all the desired codec types that should be allowed for MoH streams. By default, only G.711 mu-law is selected. To select another codec type, click on it in the scrollable list. For multiple selections, hold down the CTRL key and use the mouse to select multiple codecs from the scrollable list. The actual codec used for a MoH event is determined by the Region settings of the MoH server and the device being put on hold (IP phone, gateway, and so forth). Therefore, assign the proper Region setting to your MoH servers and configure the desired Region Relationships to control the codec selection of MoH interactions.
Note If you are using the G.729 codec for MoH audio streams, be aware that this codec is optimized for speech and it provides only marginal audio fidelity for music.
Proper IP addressing is important for configuring multicast MoH. Addresses for IP multicast range from 18.104.22.168 to 22.214.171.124. The Internet Assigned Numbers Authority (IANA), however, assigns addresses in the range 126.96.36.199 to 188.8.131.52 for public multicast applications. Cisco strongly discourages using public multicast addresses for music on hold. Instead, Cisco recommends that you configure multicast MoH audio sources to use IP addresses in the range 184.108.40.206 to 220.127.116.11, which is reserved for administratively controlled applications on private networks.
Furthermore, you should configure multicast audio sources to increment on the IP address and not the port number, for the following reasons:
IP phones placed on hold join multicast IP addresses, not port numbers.
Cisco IP phones have no concept of multicast port numbers. Therefore, if all the configured codecs for a particular audio stream transmit to the same multicast IP address (even on different port numbers), all streams will be sent to the IP phone even though only one stream is needed. This has the potential of saturating the network with unnecessary traffic because the IP phone is capable of receiving only a single MoH stream.
IP network routers route multicast based on IP addresses, not port numbers.
Routers have no concept of multicast port numbers. Thus, when it encounters multiple streams sent to the same multicast group address (even on different port numbers), the router forwards all streams of the multicast group. Because only one stream is needed, network bandwidth is over-utilized and network congestion can eventually result.
MoH Audio Sources
Configured audio sources are shared among
MoH servers in the Unified CM cluster, requiring each audio source file to be uploaded to every MoH server within the cluster. You can configure up to 51 unique audio sources per cluster (50 audio file sources and one fixed/live source via a sound card). For methods of providing additional sources, refer to the sections on Using Multiple Fixed (Live) Audio Sources, and Multicast MoH from Branch Routers.
For those audio sources that will be used for multicast streaming, ensure that
Play continuously (repeat)
are enabled. If continuous play of an audio source is not specified, only the first party placed on hold, not additional parties, will receive the MoH audio source.
Using Multiple Fixed (Live) Audio Sources
It is important to remember that only a single fixed audio source can be configured within Unified CM. However, each MoH server in the Unified CM cluster is capable of streaming a single fixed audio source by means of a Cisco MoH USB audio sound card (MOH-USB-AUDIO). When multiple fixed audio sources are needed, additional MoH servers can be added to provide these multiple sources. The audio supplied to each MoH server sound card can be the same or different, and the administrator can determine which MoH server is selected based on MRG and MRGL selections. When multiple audio sources are done in this manner, the holder's
User/Network Hold MoH Audio Source
should be configured for the fixed audio source (the single fixed audio source that is configured in Unified CM), and the MoH server to stream that fixed audio source to the device is then determined by the MRGL of the holdee.
In the case where the audio source is the same, this method also allows for redundancy of the fixed audio source. For example, in Figure 7-12 there are two MoH servers, each with an MOH-USB-AUDIO sound card connected to an audio source streaming audio derived from a live radio station feed. Phone B's MRGL contains first an MRG that contains the MOH1 server and second an MRG that contains the MOH2 server. Assuming the User/Network Hold Audio Source at Phone A has been set to the fixed audio source, after a call is established between Phone A and Phone B, and Phone B is placed on hold by Phone A, Phone B will receive the live feed audio source from the MOH1 server. In the case where the MOH1 server is down (or has no available capacity) when Phone A puts Phone B on hold, Phone B will receive the live feed audio source from the MOH2 server.
Figure 7-12 Fixed Audio Source Redundancy Example
Note Using live radio broadcasts as multicast audio sources can have legal ramifications. Consult your legal department for potential issues.
Unicast and Multicast in the Same Unified CM Cluster
In some cases, administrators might want to configure a single Unified CM cluster to handle both unicast and multicast MoH streams. This configuration might be necessary because the telephony network contains devices or endpoint that do not support multicast or because some portions of the network are not enabled for multicast.
Use one of the following methods to enable a cluster to support both unicast and multicast MoH audio streams:
Deploy separate MoH servers, with one server configured as a unicast MoH server and the second server configured as a multicast MoH server.
Deploy a single MoH server with two media resource groups (MRGs), each containing the same MoH server, with one MRG configured to use multicast for audio streams and the second MRG configured to use unicast.
In either case, you must configure at least two MRGs and at least two media resource group lists (MRGLs). Configure one unicast MRG and one unicast MRGL for those endpoints requiring unicast MoH. Likewise, configure one multicast MRG and one multicast MRGL for those endpoints requiring multicast MoH.
When deploying separate MoH servers, configure one server without multicast enabled (unicast-only) and configure a second MoH server with multicast enabled. Assign the unicast-only MoH media resource and the multicast-enabled MoH media resource to the unicast and multicast MRGs, respectively. Ensure that the
Use Multicast for MoH Audio
box is checked for the multicast MRG but not for the unicast MRG. Also assign these unicast and multicast MRGs to their respective MRGLs. In this case, an MoH stream is unicast or multicast based on whether the MRG is configured to use multicast and then on the server from which it is served.
When deploying a single MoH server for both unicast and multicast MoH, configure the server for multicast. Assign this same MoH media resource to both the unicast MRG and the multicast MRG, and check the
Use Multicast for MoH Audio
box for the multicast MRG. In this case, an MoH stream is unicast or multicast based solely on whether the MRG is configured to use multicast.
Note When configuring the unicast MRG, do not be confused by the fact that the MoH media resource you are adding to this MRG has [Multicast] appended to the end of the resource name even though you are adding it to the unicast MRG. This label is simply an indication that the resource is capable of being multicast, but the Use Multicast for MoH Audio box determines whether the resource will use unicast or multicast.
In addition, you must configure individual devices or device pools to use the appropriate MRGL. You can place all unicast devices in a device pool or pools and configure those device pools to use the unicast MRGL. Likewise, you can place all multicast devices in a device pool or pools and configure those device pools to use the multicast MRGL. Optionally, you can configure individual devices to use the appropriate unicast or multicast MRGL. Lastly, configure a User Hold Audio Source and Network Hold Audio Source for each individual device or (in the case of phone devices) individual lines or directory numbers to assign the appropriate audio source to stream.
When choosing a method for deploying both multicast and unicast MoH in the same cluster, an important factor to consider is the number of servers required. When using a single MoH server for both unicast and multicast, fewer MoH servers are required throughout the cluster. Deploying separate multicast and unicast MoH servers will obviously require more servers within the cluster.
Quality of Service (QoS)
Convergence of data and voice on a single network requires adequate QoS to ensure that time-sensitive and critical real-time applications such as voice are not delayed or dropped. To ensure proper QoS for voice traffic, the streams must be marked, classified, and queued as they enter and traverse the network to give the voice streams preferential treatment over less critical traffic. MoH servers automatically mark audio stream traffic the same as voice bearer traffic, with a Differentiated Services Code Point (DSCP) value of 46 or a Per Hop Behavior (PHB) value of EF (ToS of 0xB8). Therefore, as long as QoS is properly configured on the network, MoH streams will receive the same classification and priority queueing treatment as voice RTP media traffic.
Call signaling traffic between MoH servers and Unified CM servers is automatically marked with a DSCP value of 24 or a PHB value of CS3 (ToS of 0x60) by default. Therefore, as long as QoS is properly configured on the network, this call signalling traffic will be properly classified and queued within the network along with all other call signalling traffic.
Call Admission Control and MoH
Call admission control (CAC) is required when IP telephony traffic is traveling across WAN links. Due to the limited bandwidth available on these links, it is highly probable that voice media traffic might get delayed or dropped without appropriate call admission control. For additional information, see Call Admission Control.
Call admission control for Unified CM (based on either static locations or RSVP-enabled locations) is capable of tracking unicast MoH streams traversing the WAN but not multicast MoH streams. Thus, even if WAN bandwidth has been fully subscribed, a multicast MoH stream will not be denied access to the WAN by call admission control. Instead, the stream will be sent across the WAN, likely resulting in poor audio stream quality and poor quality on all other calls traversing the WAN. To ensure that multicast MoH streams do not cause this over-subscription situation, you should over-provision the QoS configuration on all downstream WAN interfaces by configuring the low-latency queuing (LLQ) voice priority queue with additional bandwidth. Because MoH streams are uni-directional, only the voice priority queues of the downstream interfaces (from the central site to remote sites) must be over-provisioned. Add enough bandwidth for every unique multicast MoH stream that might traverse the WAN link. For example, if there are four unique multicast audio streams that could potentially traverse the WAN, then add 96 kbps to the voice priority queue (4 ∗ 24 kbps per G.729 audio stream = 96 kbps).
Figure 7-13 shows an example of call admission control and MoH in a centralized multisite deployment. For this example, assume that IP phone C is in a call with a PSTN phone (phone B). At this point, no bandwidth has been consumed on the WAN. When phone C pushes the Hold softkey (step 1), phone B receives an MoH stream from the central-site MoH server by way of the WAN, thereby consuming bandwidth on the link. Whether or not this bandwidth is taken into consideration by call admission control depends on the type of MoH stream. If multicast MoH is streamed, then call admission control will not consider the 24 kbps being consumed (therefore, QoS on the downstream WAN interfaces should be provisioned accordingly). However, if unicast MoH is streamed, call admission control will subtract 24 kbps from the available WAN bandwidth (step 2).
Note The preceding example might seem to imply that unicast MoH should be streamed across the WAN. However, this is merely an example used to illustrate locations-based call admission control with MoH and is not intended as a recommendation or endorsement of this configuration. As stated earlier, multicast MoH is the recommended transport mechanism for sending MoH audio streams across the WAN.
Figure 7-13 Locations-Based Call Admission Control and MoH
Deployment Models for Music on Hold
The various Unified Communications call processing deployment models introduce additional considerations for MoH configuration design. Which deployment model you choose can also affect your decisions about MoH transport mechanisms (unicast or multicast), resource provisioning, and codecs. This section discusses these issues in relation to the various deployment models.
For more detailed information about the deployment models, see the chapter on Collaboration Deployment Models.
Single-Site Campus (Relevant to All Deployments)
Single-site campus deployments are typically based on a LAN infrastructure and provide sufficient bandwidth for large amounts of traffic. Because bandwidth is typically not limited in a LAN infrastructure, Cisco recommends the use of the G.711 (A-law or mu-law) codec for all MoH audio streams in a single-site deployment. G.711 provides the optimal voice and music streaming quality in an IP Telephony environment.
MoH server redundancy should also be considered. In the event that an MoH server becomes overloaded or is unavailable, configuring multiple MoH servers and assigning them in preferred order to MRGs ensures that another server can take over and provide the MoH streams.
With the increasing diversity of network technologies, in a large single-site campus it is likely that some endpoint devices or areas of the network will be unable to support multicast. For this reason, you might have to deploy both unicast and multicast MoH resources. For more information, see the section on Unicast and Multicast in the Same Unified CM Cluster.
To ensure that off-net calls and application-handled calls receive expected MoH streams when placed on hold, configure all gateways and other devices with the appropriate MRGLs and audio sources, or assign them to appropriate device pools.
Centralized Multisite Deployments
Multisite IP telephony deployments with centralized call processing typically contain WAN connections to multiple non-central sites. These WAN links usually cause bandwidth and throughput bottlenecks. To minimize bandwidth consumption on these links, Cisco recommends the use of the G.729 codec for all MoH audio streams traversing the WAN. Because the G.729 codec is optimized for voice and not music applications, you should use G.729 only across the WAN, where the bandwidth savings far outweighs the lower quality afforded by G.729 for MoH transport. Likewise, because multicast traffic provides significant bandwidth savings, you should always use multicast MoH when streaming audio to endpoints across the WAN.
If the sound quality of an MoH stream becomes an issue when using the G.729 codec across the WAN, you can use the G.711 codec for MoH audio streams across the WAN while still using G.729 for voice calls. In order to send MoH streams across the WAN with the G.711 codec but voice calls across the WAN with the G.729 codec, place all MoH servers in a Unified CM region by themselves, and configure that region to use G.711 between itself and all other regions. Thus, when a call is placed between two phones on either side of a WAN, the G.729 codec is used between their respective regions. However, when the call is placed on hold by either party, the MoH audio stream is encoded using G.711 because G.711 is the configured codec to use between the MoH server's region and the region of the phone placed on hold.
Multicast MoH from Branch Routers
Branch routers deployed with the Cisco Unified Survivable Remote Site Telephony (SRST) feature can provide multicast MoH in a remote or branch site, with the MoH streaming from the branch SRST router's flash or from a live feed connected to an analog port. Multicast MoH from a branch router via these two methods enhances the Cisco Unified Communications MoH feature in both of the following scenarios:
When the WAN is up and the phones are controlled by Unified CM, this configuration can eliminate the need to forward MoH across the WAN to remote branch sites by providing locally sourced MoH.
When SRST is active and the branch devices have lost connectivity to the central-site Unified CM, the branch router can continue to provide multicast MoH.
When using the live feed option in either scenario, the SRST router provides redundancy by monitoring the live feed input, and it will revert to streaming MoH from a file in flash if the live feed is disconnected. You can use only a single multicast address and port number per SRST router to provide multicast MoH; therefore, the SRST router does not support streaming from both the live feed and the flash file at the same time. In addition, the SRST router can stream only a single audio file from flash.
Note An SRST license is required regardless of whether the SRST functionality will actually be used. The license is required because the configuration for streaming MoH from branch router flash is done under the SRST configuration mode and, even if SRST functionality will not be used, at least one max-ephones and one max-dn must be configured.
During non-fallback mode (when the WAN is up and SRST is not active), the branch SRST router can provide multicast MoH to all local Cisco Unified Communications devices. To accomplish this, you must configure a Unified CM MoH server with an audio source that has the same multicast IP address and port number as configured on the branch router. In this scenario, because the multicast MoH audio stream is always coming from the SRST router, it is not necessary for the central-site MoH server audio source to traverse the WAN.
To prevent the central-site audio stream(s) from traversing the WAN, use one of the following methods:
Configure a maximum hop count
Configure the central-site MoH audio source with a maximum hop count (or TTL) low enough to ensure that it will not stream further than the central-site LAN.
Configure an access control list (ACL) on the WAN interface
Configure an ACL on the central-site WAN interface to disallow packets destined to the multicast group address(es) from being sent out the interface.
Disable multicast routing on the WAN interface
Do not configure multicast routing on the WAN interface, thus ensuring that multicast streams are not forwarded into the WAN.
Figure 7-14 illustrates streaming multicast MoH from a branch router when it is not in fallback mode. After phone A places phone C on hold, phone C receives multicast MoH from the local SRST router. In this example, the MoH server is streaming a multicast audio source to 18.104.22.168 (on RTP port 16384); however, this stream has been limited to a maximum hop of one (1) to ensure that it will not travel off the local MoH server's subnet and across the WAN. At the same time, the branch office SRST router/gateway is multicasting an audio stream from either flash or a live feed. This stream is also using 22.214.171.124 as its multicast address and 16384 as the RTP port number. When phone A presses the Hold softkey, phone C receives the MoH audio stream sourced by the SRST router.
Figure 7-14 Multicast MoH from Branch Router
When using this method for delivering multicast MoH, configure all devices within the Unified CM cluster to use the same user hold and network hold audio source and configure all branch routers with the same multicast group address and port number. Because the user or network hold audio source of the holder is used to determine the audio source, if you configure more than one user or network hold audio source within the cluster, there is no way to guarantee that a remote holdee will always receive the local MoH stream. For example, suppose a central-site phone is configured with an audio source that uses group address 126.96.36.199 as its user and network hold audio source. If this phone places a remote device on hold, the remote device will attempt to join 188.8.131.52 even if the local router flash MoH stream is sending to multicast group address 184.108.40.206. If instead all devices in the network are configured to use the user/network hold audio source with multicast group address 220.127.116.11 and all branch routers are configured to multicast from flash on 18.104.22.168, then every remote device will receive the MoH from its local router.
In networks with multiple branch routers configured to stream multicast MoH, this allows for more than 51 unique MoH audio sources within the Unified CM cluster. Each branch site router can multicast a unique audio stream, although all routers must multicast this audio on the same multicast group address. The MRGL assigned to the gateway determines the MoH server used for MoH streaming when a PSTN caller is placed on hold, while the phone placing the call on hold determines the audio source. You can provide more than 51 unique MoH streams for locations in excess of 51 by providing enough MoH servers and therefore more than 51 multicast addresses. The following example illustrates how to deploy MoH streaming from more than 51 local branch gateways:
Branch gateways 1 to 51 are configured with an MRGL pointing to MoH server node MoH_1 (with base multicast address of 22.214.171.124), with all phones at each branch pointing to one of the 51 audio sources configured on the cluster. Thus, phones in Branch 1 point to audio source 1, which on MoH_1 server is 126.96.36.199 to 4 (depending on codec and assuming audio sources are configured in order); phones in Branch 2 point to audio source 2, which on MoH_1 server is 188.8.131.52 to 8; phones in Branch 3 point to audio source 3, which on MoH_1 server is 184.108.40.206 to 12; and so on up to phones in Branch 51 pointing to audio source 51, which on MoH_1 server is 220.127.116.11 to 204.
Branch gateways 52 to 102 are configured with an MRGL pointing to MoH server node MoH_2 (with base address 18.104.22.168), with all phones at each branch pointing to one of the same 51 audio sources configured on the cluster. In this case, however, phones in Branch 52 point to audio source 1, which on MoH_2 server is 22.214.171.124 to 4 (depending on codec and assuming audio sources are configured in order); phones in Branch 53 point to audio source 2, which on MoH_2 server is 126.96.36.199 to 8; phones in Branch 54 point to audio source 3, which on MoH_2 server is 188.8.131.52 to 12; and so on up to phones in Branch 102 pointing to audio source 51, which on MoH_2 server is 184.108.40.206 to 204.
However, in a centralized PSTN deployment with a single gateway or set of gateways for PSTN access, there is no way to configure more than 51 unique audio sources. With a centralized PSTN deployment, because there are no PSTN gateways at the local sites, you cannot use an MRGL to point to multiple MoH servers based on branch location. Thus, in a centralized PSTN deployment you can have at most 51 unique site-specific MoH sources or multicast streaming addresses enabling unique MoH sources for only 51 branch sites.
During fallback mode (when the WAN is down and SRST is active), the branch SRST router can stream multicast MoH to all analog and digital ports within the chassis, thereby providing MoH to analog phones and PSTN callers.
The branch router's configuration for fallback mode multicast MoH is the same as the normal operation configuration. However, which multicast address you configure on the router depends on the intended operation. If you want the branch router to provide multicast MoH to devices only in fallback mode (for example, if MoH received by remote devices is to be sourced from the central-site MoH server during non-fallback mode), then the multicast address and port number configured on the SRST router should not overlap with any of the central-site MoH server audio sources. Otherwise, remote devices might continue to receive MoH from the local router flash, depending on the configured user/network hold audio sources.
Note that, once the branch SRST/gateway router is configured to provide multicast MoH, the router will continue to multicast the MoH stream even when not in fallback mode.
It is also possible to configure the fallback mode to use Cisco Unified Communications Manager Express (Unified CME) in Enhanced SRST mode. Fallback mode behavior is still the same, but the configuration commands are slightly different. SRST commands are entered under the Cisco IOS
construct, while the commands for Enhanced SRST are entered under
There are four methods of providing multicast MoH via SRST:
SRST multicast MoH from branch router flash
SRST multicast MoH from a live feed
Enhanced SRST with multicast MoH from branch router flash
Enhanced SRST with multicast MoH from a live feed
For more details on configuration of Cisco Unified SRST and Enhanced SRST, refer to the following documentation:
Cisco Unified SRST System Administrator Guide
, available at
Cisco Unified Communications Manager Express System Administrator Guide
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Distributed Multisite Deployments
Multisite IP telephony deployments with distributed call processing typically contain WAN or MAN connections between the sites. These lower-speed links usually cause bandwidth and throughput bottlenecks. To minimize bandwidth consumption on these links, Cisco recommends use of the G.729 codec for all MoH audio streams traversing them. Because the G.729 codec is optimized for voice and not music applications, you should use G.729 only across the WAN/MAN links, where the bandwidth savings far outweighs the lower quality afforded by G.729 for MoH transport.
Unlike with centralized multisite deployments, in situations where G.711 might be required for MoH audio streams traveling across a WAN, MoH audio streams cannot be forced to G.711 in a distributed multisite deployment. Even when MoH servers are placed in a separate Unified CM region and the G.711 codec is configured between this region and the intercluster or SIP trunk's region, the codec of the original voice call is maintained when a call between the two clusters is placed on hold by either phone. Because these intercluster calls are typically encoded using G.729 for bandwidth savings, a MoH stream from either cluster will also be encoded using G.729.
Another option is to provision multicast MoH for intercluster calls across an intercluster trunk (ICT) or SIP trunk. This allows endpoints in one Unified CM cluster to hear multicast MoH streamed from another Unified CM cluster, while making more efficient use of intercluster bandwidth. A properly designed IP Multicast environment is required to take advantage of this feature. For more information on IP Multicast, refer to the documentation available at
Proper multicast address management is another important design consideration in the distributed intercluster environment. All MoH audio source multicast addresses must be unique across all Unified CM clusters in the deployment to prevent possible overlap of streaming resources throughout the distributed network.
Clustering Over the WAN
As its name suggests, clustering-over-the-WAN deployments also contain the same type of lower-speed WAN links as other multisite deployments and therefore are subject to the same requirements for G.729 codec, multicast transport mechanism, and solid QoS for MoH traffic traversing these links.
In addition, you should deploy MoH server resources at each side of the WAN in this type of configuration. In the event of a WAN failure, devices on each side of the WAN will be able to continue to receive MoH audio streams from their locally deployed MoH server. Furthermore, proper MoH redundancy configuration is extremely important. The devices on each side of the WAN should point to an MRGL whose MRG has a priority list of MoH resources with at least one local resource as the highest priority. Additional MoH resources should be configured for this MRG in the event that the primary server becomes unavailable or is unable to process requests. At least one other MoH resource in the list should point to an MoH resource on the remote side of the WAN in the event that resources at the local side of the WAN are unavailable.