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.

Converting Voice Signal Into VoIP

For the transmission of the voice signal over IP network, the analog voice signal must be converted to the digital VoIP packets. For that the voice signal must be pass through following steps in order: Sampling, Quantization, Encoding, and optionally Compression. 

Sampling

Sampling is the process of chopping analog signal into regular intervals.Harry Nyquist prove that if the signal is sampled at the rate of twice the highest frequency (2*4000 kHz), the samples will contain sufficient information to accurately reconstruct signal at the receiver end.The figure below shows the sampling of an analog signal.

Sampling

• The human ear can sense sounds from 20 to 20,000 Hz
• Human speech uses frequencies from 200 to 9000 Hz
• The telephone channel was designed to operate at frequencies of 300 to 4000 Hz

Quantization 

Quantization divides  an analog signal sample into a set of discrete steps that are closest in value to the original analog signal.

The amplitude range is divided into 16 segments (0 to 7 positive, and 0 to 7 negative). Starting with segment 0, each segment has less-granular intervals than the previous segment, which reduces the signal-to-noise ratio (SNR) and makes the segment uniform.The sample value ranges from +127 to -127.

Quantization

Encoding

Encoding convert decimal values in the binary expressions of either 1 and 0. After sampling the signal next step to encode samples for the transmission over the telephony network.This process is called pulse-code modulation(PCM).  The PCM process, as shown in Figure below.

Coding

•  The first bit (MSB) identifies polarity
•  Bits two, three, and four identify segment
•  Final four bits quantize the segment

In the United States, Canada, and Japan, mu-law is used. The rest of the world uses a-law.

Compression

Signal compression is used to reduce the bandwidth usage per call. The most common codec algorithms are presented in the table below.

Compression

Summery

These four steps are performed by the DSP {Digital Signal Processor} at the originating gateway.The VoIP packets are then traveled to the destination gateway. DSPs on the destination side decodes the payload and reverse the process performed on the originating gateway.

Exam Resources and Books for CCNA Collaboration

In the world of Cisco the main resources for exam preparation for  CCNA collaboration is Cisco's Official Certification Guide.I have shared some links from where you get and download PDF copy of books.You will also find the topics for both exams.

CICD Exam Topics

The official study guide helps you master topics on the CCNA Collaboration CICD 210-060 exam, including the following:

• Cisco Unified Communications components
• Cisco Unified Communications Manager Express administration, end user management, dial plans, and telephony features
• Cisco Unified Communications Manager administration, end point management, dial plan elements and interactions, and telephony and mobility features
• Cisco Unity Connection voicemail
• CM IM and Presence support
• CME and CUCM management and troubleshooting
• Monitoring Cisco Unity Connection

You can also download  topics from Here.

CICD Books

• CCNA Collaboration CICD Official Cert Guide available Here.
• You can also buy from Amazon.
• You can also download PDF which I have shared Here.

CIVND Exam Topics

The official study guide helps you master topics on the CCNA Collaboration CIVND 210-065 exam, including the following:

• Cisco Collaboration components and architecture
• Cisco Digital Media Suite, Digital Signs, Cisco Cast, and Show and Share
• Cisco video surveillance components and architectures
• Cisco IP Phones, desktop units, and Cisco Jabber
• Cisco TelePresence endpoint portfolio
• Cisco Edge Architecture including Expressway
• Multipoint, multisite, and multiway video conferencing features
• Cisco TelePresence MCU hardware and server family
• Cisco TelePresence management
• Cisco WebEx solutions

You can also download  topics from  Here.

CIVND Books

• CCNA Collaboration CIVND Official Cert Guide available Here.
• You can also buy from Amazon.
• You can also download PDF which I have shared Here.

My Approach to CCNA Collaboration Certification

In this my first blog I'll drive you straight to how to make your self ready for getting Cisco Collaboration certificate in  you first attempt.

Cisco Collaboration has two exams to pass.

1.   210-060 CICD
2.  210-065 CIVND

You will find exam topics and policies over there.

For getting certified you should have basic knowledge of CCNA R&S.It will helpful to get this smoothly done. VLAN, DHCP, Switching ,Subnetting concepts are need to be more clear before you jump in Collaboration world ,though Cisco does not require to have any prerequisite.

For the preparation I recommend to use Cisco official Cert Guides for both CICD and CIVND. These two books are enough to get certified.

First you have to pass CICD and then go for CIVND.Each Exam contains 55-65 questions.But mostly 60 questions are there.Exam passing score is 860 for now it may vary timely.You will be given 90 minutes to finish exam.The questions are of multiple choice,drag n drop.You can book your exam from Pearson Vue. 

The exam cost 250 USD.This amount is for single time exam.If you unable to clear exam you have to pay fees again.So be careful and have your exam when your are confident enough.

Only theory knowledge is not enough to crack exam you'll also need to have strong practical knowledge of CUCM. There also comes few other components like CME CUC,Jabber.Talk on these components in my upcoming blogs.CUCM is the king in Collaboration world.

In my next blog You will find available resources for certification and more toward CICD topics.If you have any question please write in comment.