CUCM:Network and Feature Services

Network Services

Network services can not be activated or deactivated.But they can only be started, stopped or restarted by the network admin from Cisco Unified Serviceability web interface.Network services are automatically activated from Cisco Unified Serviceability path, Tools > Control Center > Network Services.

Some of the examples of network services are as below.

• CDR Services: Cisco CDR Repository Manager
• Admin Services: Call Manager Admin
• CM Services: Call Manager IP Phone Service, Call Manager Extension Mobility Services
• Platform Services: Cisco Tomcat, Cisco Database, Cisco SNMP Agent
• Back up and Restore Services: Cisco CallManager Serviceability, Cisco CDP, Cisco Trace Collection Service
• Performance and Monitoring Services: RTMT Service, Cisco RTMT Reporter

Feature Services

Feature services can be activated or deactivated on per server basis.Feature Services can also be started, stopped or restarted by network admin from Cisco Unified Serviceability web interface.For that choose path ,Tools > Control Center > Network Services.

These feature services can also be activated or deactivated from Serviceability web interface path,

Tools > Service Activation.

Some of the examples of feature services are as below.

• Directory Service: Cisco Dirsync
• CM Service: Cisco CallManager, Cisco TFTP, Cisco CTIManager
• Performance and Monitoring Service: Cisco Serviceabilty Reporter
• CTI Services: Cisco CM Attendant Console Server,Cisco IP Manager Assistant
• Database and Admin Service: Cisco AXL Web Service ,Cisco TAPS Service,Cisoc Bulk Provisoning Service

User Facing Features in CUCM

In the CUCM cluster the publisher is the only server which contains read/write copy of database.All the configuration changes are done on the publisher server.Publisher server replicates the read only copy of changes to all the subscribers in the cluster.

The dynamic information which can be modified during the outage of the publisher is known as User Facing Features (UFF).

UFF data is replicated between all the servers in clusters.

Some of the examples of UFFs are as below.

• Message Waiting Indicator (MWI)
• Hunt group login status
• Device mobility (DM)
• Do Not Disturb, Enable/Disable (DND)
• Privacy, Enable/Disable
• Extension Mobility Login (EM)
• Call Forward All (CFA)

Timers in SNR Or Unified Mobility

Single number reach [SNR] or Unified Mobility allows the user to pick up enterprise call to his/her mobile phone when user not available in office.If user is back in office while call is going on, its allowed user to switch the phone from mobile to office desk phone.

How it works?

1. Assign IP phone soft-keys in two stages

• On hook
• Connected

2. Create End User in User Management >End User

• Check Enable Mobility to Enable 

3. Associate user to the IP Phone.

4. Add remote destination profile in Device > Device Settings > Remote Destination Profile > Add new. Click Save, now you can see an option to add a new Directory number (DN) .

5. Assign Directory Number and Rerouting CSS here.

6. At the end of this page click on Add a New Remote Destination and add remote destination number or mobile number here.

At the bottom of this page you will find Timer Information.These timers are listed below.

Answer Too Soon Timer

The minimum time in millisecond that CUCM requires the mobile phone to ring before answering the call.This settings accounts for the situations where the mobile phone is switched off or not reachable, in which network may immediately divert call to mobile phone voice mail.If the mobile phone is answered before this timer is expired,CUCM pulls call back to the enterprise number.

Range: 0-10,000 ms

Default:1 5,000 ms

Answer Too Late Timer

It is the maximum time in millisecond that CUCM allow to ring mobile phone.If this value is reached CUCM stop ringing mobile phone and pulls the call back to the enterprise number.

Range: 0 & 10,000-300,000 ms

Default: 19,000 ms

Delay Before Ringing Timer

Time that elapse before the mobile phone rings when call is extended to reach to remote destination.

Range: 0-30,000 ms

Default: 4,000 ms

CUCM:Line CSS & Device CSS

Before going ahead I hope you know the terms CSS and Partitions very well.If not you take a look at here.

Device CSS is applied to the IP phone added in the CUCM.Line CSS is applied to the Line / directory number.

If both CSS applied together then line CSS get higher priority and device CSS get second priority.

Cisco recommends to apply route pattern to the device and apply block pattern to the line CSS.

Here we take an example of making Local,Long distance,International and emergency calls.
We have PHONE_A with line number 1001. PHONE_A is allowed to call Local Numbers  and Emergency calls only.

For that we need to create two CSS. One in which contains all patterns  and in second we keep only blocked patterns.

CSS_Allow_ALL contains following partitions.

PT_Local
PT_Internation
PT_Longdistance
PT_Emergency

For the blocking we keep blocked numbers : Long distance partition and International partition in
CSS_Block.

CSS_Block contains following partitions.

PT_Internation
PT_Longdistance

So, Here we applied CSS_Allow_ALL to PHONE_A's device CSS and CSS_Block to PHONE_A's line CSS.With that PHONE_A is allowed to call only Local Numbers and Emergency calls only.

SIP Early Offer vs Delayed Offer

Early Offer
Initial SIP INVITE is sent with SDP message body. Session initiator allows called device to choose its codec for the session.

Why early media?

The called device might want to establish an early media RTP path to reduce the effect of audio cut-through delay for calls experiencing a long signal delay.

Calling device might want to establish early media RTP path to access DTMF or Voice driven IVR system.

Delayed Offer

Initial SIP message is sent without SDP message body.Session initiator waits for the called device to send capabilities first.

Delayed offer is recommended for SIP trunks.By default, CUCM supports delayed offer. 

CUCM - Partition and Calling Search Space

Partition and CSS are calling privileges control elements which are use to block or allow user by making certain calls.

Partition

A partition is group is any dialable patterns with same accessibility.Any dilable entity can be added in partition.By default all numbers get <none> partition.

CSS

Calling Search Space is ordered list of partitions.When calling side CSS contains called side partition then call is connected.All the devices by default get <none> CSS.You can not make changes in <none> CSS.By default at the end of each CSS there is a <none> partition,which is invisible.

Media Resources In CUCM

Media resource is software or hardware based entity that performs media processing functions on the data stream to which it is connected.

In Cisco Unified Communication Manager IP VMS [IP Voice Multimedia Streaming Service] is used to provide software based media resources. Digital signal processors are used to provide both hardware and software based media resources.

Conference Bridge [CFB]

Conference Bridge mix the audio streams from all the participants in the conference.It collects the audio from all participants but doesn't send back to the originating device.

CFB is both in hardware and software based media resources.

If all the participants in the conference are using G.711 mu-law then software based media resources are used.

If all participants use a codec other than G.711 mu-law then hardware based media resources are used.

Media Termination Point [MTP]

Media Termination Point is an entity that bridge two full duplex media stream together and allow them to set up and tear down independently.

It is used for transcoding from  G.711 mu-law to G.711 a-law and vice  versa. 

CFB is both in hardware and software based media resources.

A software MTP supports only G.711 streams or passthrough mode in the codec.

A hardware MTP is used to bridge two connections which utilized different packetization period.

Annunciator

An annunciator is a software based media resource which provides the ability to stream messages or call processing tones from system to user.It uses SCCP  messages to establish one-way RTP streams.

For most SIP devices call processing tones are downloaded at the time of registration.

To establish two-way media connection with an annunciator enable/true Duplex Streaming in CUCM service parameter.

An annunciator is capable of supporting G.711 a-law, G.711 mu-law, G.729, Cisco wideband codecs without transcoding.

It is automatically created when IP VMS service is activated in CUCM.

It registers with the single CUCM at a time as defined by its device pool and CM group.

An annunciator supports 48 simultaneous streams if it is running on the same server with the CM services. 

Music On Hold [MoH]

MoH is a software based media resource which provides one-way music stream to the user which is put on hold.MoH server must share the following information with the CM cluster through the DB replication process.

• Audio source
• Unicast or Multicast 
• Multicast Address 

It is required to enable IP VMS service on all CM nodes to configure MoH server.

CUCM supports unicast and multicast MoH transport mechanism.

Unicast MoH

It is a point -to point one-way audio stream between MoH server to end points.Unicast MoH create a separate stream for each user/endpoint.

Unicast MoH call flow is initiated by a message from Unified CM to MoH server.

Multicast MoH

It is a point-to-multipoint one-way RTP stream between MoH server and multicast group IP address.The endpoints requesting MoH can join a group as needed.

Multicast MoH is initiated by a message from Unified CM to a holdee device. This message instruct device to join the multicast group address of the configured MoH audio stream.

Type of Hold

Two types of Holds are there.

• User Hold - User hold at IP phone or at PSTN
• Network hold- Network hold can be occurred  from Call Park, Call Transfer, Conference  or Application based hold

Holder and Holdee

The basic process of MoH in CUCM consist of two components, they are holder and holdee.Holder is end point user or device which place a call on hold. Holdee is end point user or device which is placed on hold.

Understanding Cisco Dial Peers

CME router can be connected to many numbers of analog and digital connections.But this router will not know how to use it unless you define or create dial-peers for those connections.

Dial peers are the entity that defines voice reachability information.Dial peers established logical connections, called call legs, to complete the end-to-end call.

Dial peers are generally classified into two categories.

POTS dial Peers: Provides connection or voice reachability information for traditional telephony networks like PSTN, PBX, analog telephones.It also includes devices connected to the FXO, FXS and  E&M.
POTS dial-peer includes devices those doesn't have an IP address like analog phones, Fax, PBX, PSTN.

VoIP dial Peers: Provides voice reachability information of VoIP networks. Points to the IP address or DNS name of the destination VoIP device that terminates the call.VoIP dial peers map a dial string to a remote network device.The remote devices are CUCM, CME, SIP proxy, H.323 gateway, any voice gateway.

Configuring POTS dial Peers
Figure 1.1 blow shows network for dial-peers configuration. The configuration of POTS dial-peers begins with the CME A.Two analogs phones are connected to the CME A route via FXS ports.

Figure 1.1: Dial-peer Configuration Network

There are main two parameters in POST dial-peer which are needed to be specified.They are the telephone number and voice port. 

Example 1-1: POTS dial-peer Configuration

As shown in the example 1-1 to create dial-peer you need to enter into global configuration mode of the router CME A. In global configuration mode use command dial-peer voice tag pots.Where the tag is any number you want to give. It is not necessary to have the same tag number and destination-pattern number.

The destination-pattern is used for matching of called telephone number. In this example, we have number "1101".

The port command specifies the respective voice port. In this example, port 0/0/0 defines that the port is on module 0(1st), voice interface card (VIC) slot 0 (2nd), and voice port 0.

In POTS dial-peer voice port automatically strips explicitly defined dial numbers.To prevent this in the router no digit-strip command is used.The forward-digits all command is used on a router to forward all defined digits.

Example 1-2: POTS dial-peer Configuration

Configuring VoIP dial peer

As shown in figure CME A and ROUTER B are connected via IP WAN.TO make calls for this type of connectivity you must use VoIP dial-peers because the call is crossing an IP connectivity of the network.

Example 1-3: VoIP dial-peer Configuration

As shown in example 1-3 above, in VoIP dial-peer, compared to POTS session-target is used instead of tag.This command is often used with the syntax session-target ipv4:ip_address, where IP is of remote entities like CUCM, CME, Gateways etc.
The codec value is set by using codec command.The default codec value for VoIP dial-peer  is G.729.
Mismatch in codec value between two routers results in call failure and returns fast busy signal.

VoIP dial-peer does not strip explicitly defines numbers.So no need to define no digit-strip or  forward-digits all command.

In the next blog you find more about dial peer wild card and dial patterns.

Cisco IP Phone Boot Process

Once IP phone is connected to a network, it goes to following standard steps to get registered to Call Manager and to get a  directory number. A figure below shows the overview of Cisco IP phone startup process when IP phone is connected to Cisco Catalyst switch, capable of providing PoE.

IP Phone Startup process

1. Obtain power from the switch: When an IP phone is connected to Cisco Catalyst switch, it is able to provide in-line power or Power Over Ethernet.The switch will first send Fast Link Pulse (FLP) signal.The switch will use FLP to determine if attached phone is unpowered IP phone.If a unpowered state is determined, Cisco IPphone loops back FLP, signalling the switch to send -48 V DC power down the line.

2. Load the stored phone image: The Cisco IP Phone has non-volatile Flash memory in which it stores firmware images and user-defined preferences. At startup, the phone runs a bootstrap loader that loads a phone image stored in Flash memory. Using this image, the phone initializes its software and hardware

3. Configure VLAN: After receiving power and loading image file in IP Phone, a switch will send the Cisco Discovery Protocol (CDPv2) trigger packet to the IP phone.This CDP packet provides VLAN information to the IP phone.The IP phone will then tag all the with appropriate Voice VLAN information.

4. Obtain IP address and TFTP server address: As the IP phone learn VLAN information the IP Phone broadcasts a request to a DHCP server. The DHCP server responds to the IP Phone with an IP address, a subnet mask, and the IP address of the Cisco TFTP server.

DHCP server can provide the location of TFTP server to the Cisco IP phones in two different ways.

1. DHCP option 66: It gives the Cisco IP phone the hostname of the TFTP server.
2. DHCP option 150: It gives the Cisco IP phone the IP address of the TFTP server.

5. Contact TFTP server for configuration: The TFTP server has configuration files for  IP phones, which define parameters for connecting to Cisco CallManager. The TFTP server sends the configuration information for that IP Phone, which contains an ordered list of up to three Cisco CallManagers.

6. Register with Cisco CallManager: After obtaining the file from the TFT P server, the phone attempts to make a TCP connection to the highest priority Cisco CallManager on the list.After registration, it receives extension number and become operational.