CICD - Descriptions Flashcards
UNITY - Cisco Unity Connection (CUC) - 7 Keys
- Voicemail, auto-attendant, voice recognition (VUI)
- Max 20k mail boxes per CUC server
- Integrated IMAP server (Email Client Access) or Exchange Integration
- Web-based Voicemail Access
- Integrated with SCCP or SIP signaling and Traditional Telephony (PBX)
- Visual voicemail (Phone view) with Jabber or Cisco IP phones
- CUC needs a SIP TRUNK to be Integrated with CUCM
Cisco Unity Connection Architecture
Cisco Unity Express - 300 per Server - Router - NO
Unified CM Business Edition - 500 per Server - Appliance - NO
Cisco Unity Connection - 20,000 per Server - Appliance - Active / Active
Cisco Unity - 15,000 per Server - Windows Server - Active / Passive
Cisco Unity Connect or Connection has a different type of storage of your messages than Cisco Unity. Cisco unity actually needs an external e-mail solution such as Exchange or Lotus Notes because it leverages the exchange database store to actually store your voicemail messages with Cisco Unity Connection all of the messages are stored locally in an IBM Informix Database. So, we don’t have to rely upon, in other words that exchange server to be up and running in order for our messages to be stored
IMP - Instant Messaging and Presence - 5 Keys
- Cisco version of HANGOUTS, SKYPE, SAMETIME
- Integrated with CUCM (2 modes)
- Allows connections to outside domain (WEBEX, GOOGLE TALK,…)
- Prefers JABBER
- Max 45k users in UC mode
Max 75k users in STANDALONE mode
Max 40k users in LYNC interop mode (Skype for Business)
TMS - Telepresence Management Suite - 6 Keys
- Telepresence device provisioning
- Outlook (Exchange) telepresence scheduling
- Click-to-call video conferences
- Multi-source phone books
- Management of in-progress conferences
- Ressource utilization reports
VCS - Video Communication Server - 5 Keys
- Is like CUCM to telepresence endpoints
- Can handle video calls for registered endpoints
- VCS (combined with TMS) handles “UNCONTROLED” endpoints
- Includes : VCS CONTROL (incoming and outgoing calls to video endpoints)
VCS EXPRESSWAY (firewall traversal through “CORE” and “EDGE” components) - Can enable the “JABBER GUEST”
CUBE - Cisco Unified Border Element
Is designed to be the marker between your network and someone else network. CUBE is like your firewall for voice. This is the end of your company network. CUBE will be the only known server by your ITSP and will, based on Dial Plan, deliver to your company servers
VoIP Provider Connect
In order for us to connect out to let’s say an IP telephony service provider for example, we would need a special operating system and that special operating system is called the Cisco Unified Border Element. We call it CUBE for short, and what this allows us to do is connect a Voice over IP connection to another Voice over IP connection. In this example we’re talking about service provider, so if I want to connect my network to an IP telephony service provider, I would need to use this Cisco Unified Border Element to interconnect these two networks, and basically what happens here is the CUBE helps us with a demarcation point between the two entities, so between myself and the service provider we can have security, NAT - I may not want my private IP addresses exposed and then we might even want billing to take place.
JABBER - the ultimate collaboration clients - 6 Keys
- All plateformes Softphone, can Control desktop IP phones (can also replace the desktop Phone)
- Instant Messaging and Presence
- Employee Directory (LDAP)
- Voicemail Control
- Audio and Video Calls / Web Conferencing
- Communication History
Cisco Jabber client applications
These Cisco Jabber client applications reside on top of the Clients Services Framework, which provides a simplified client interface and an abstraction layer that allows access to the following underlying communications services:
- SIP-based call control for voice and video softphone clients from UnifiedCM
- Deskphone call control and “Click to Dial” services from UnifiedCM’s CTI interface
- Voice and video media termination for softphone clients
- Instant messaging and presence services using XMPP, from either the Cisco IM and Presence Service or Cisco WebEx. Cisco WebEx Meeting Center also offers hosted collaboration services such as online meetings and events
- Scheduled audio, video and web conferencing services
- Desktop sharing using either, video desktop sharing (BFCP) or WebEx desktop sharing
- Visual voicemail services from Cisco Unity Connection using IMAP
•Contact management using:
–UnifiedCM User Data Service (UDS) as a contact source (LDAP directory synchronization supported)
–Directory access using Microsoft Active Directory or supported LDAP directories as a contact source
–WebEx Messenger service
–Client Services Framework cache and contact list
•Microsoft Office Integration, which provides user availability status and messaging capabilities directly through the user interface of Microsoft Office applications such as Microsoft Outlook
VLAN - Virtual Local Area Network - 5 Keys
- Logically Groups Users
- Segments Broadcast Domain
- Subnet Correlation
- Access Control
- Quality of Service
VOICE VLAN - 4 Keys
- TRUNK = TAGGING PORT
- Phone connected to a type of “TRUNK” port
- Phone sends TAGGED TRAFFIC
- Computer sends UNTAGGED TRAFFIC
Data traffic needs to be segmented into 4 types of Data
- Mission critical (business Data)
- Transactional (dB replication)
- Best Effort (web surfing)
- Scavenger (peer to peer traffic)
QoS - MODELS
- Best effort
- Integrated Service (INTSERV) based on RSVP
- Differenciated Service (DIFFSERV)
QoS - METHODS
- Classification and marking
- Queuing (congestion management)
- Policing and Shapping
- Congestion avoidance (WRED)
- Link efficiency
QoS - Queuing Algorithms (5)
FIFO
First packet in is first packet out; only one queue
Priority queuing (PQ) Empty queue 1l if queue 1 is empty, then dispatch packets from queue 2, and so on
Weighted fair queuing (WFQ)
Flow-based algorithm that simultaneously schedules interactive traffic to the front of a queue
Class-based weighted fair queueing (CBWFQ)
Extends WFQ functionality to provide support for user-defined traffic classes
Low-latency queueing (LLQ)
Brings strict PQ to CBWFQ; allows delay-sensitive data like voice to be dequeued and transmitted before packets in other queues are dequeued
QoS - Link Efficiency Mechanisms
- PAYLOAD COMPRESSION
Squishing the data - HEADER COMPRESSION
Squishing the header (key for RTP) - LINK FRAGMENTATION AND INTERLEAVING
Blowing up big packets
DSCP and IP PRECEDENCE
DSCP IP Precedence Conversion Table
At its simplest, the higher the value of the IP Precedence field, the higher the priority of the IP packet. Simple…
DSCP Name - DS Field Value in Binary/Decimal - IP Precedence CS0 - 000 000/0 - 0 CS1 - 001 000/8 - 1 AF11 - 001 010/10 - 1 AF12 - 001 100/12 - 1 AF13 - 001 110/14 - 1 CS2 - 010 000/16 - 2 AF21 - 010 010/18 - 2 AF22 - 010 100/20 - 2 AF23 - 010 110/22 - 2 CS3 - 011 000/24 - 3 AF31 - 011 010/26 - 3 AF32 - 011 100/28 - 3 AF33 - 011 110/30 - 3 CS4 - 100 000/32 - 4 AF41 - 100 010/34 - 4 AF42 - 100 100/36 - 4 AF43 - 100 110/38 - 4 CS5 - 101 000/40 - 5 EF - 101 110/46 - 5 CS6 - 110 000/48 - 6 CS7 - 111 000/56 - 7
CS - Class Selector (RFC 2474)
AFxy - Assured Forwarding (x=class, y=drop precedence) (RFC2597)
EF - Expedited Forwarding (RFC 3246)
Where can you apply QoS methods For INBOUND QoS - 3 Keys
- Classify
- Mark
- Police
Where can you apply QoS methods For OUTBOUND QoS - 7 Keys
- Queue
- Mark
- Avoid
- Shape
- Police
- Compress
- LFI
AUTOQoS VOIP / AUTOQoS DISCOVERY… - 5 Keys
- Provides a template based QoS configuration
- Reduce QoS deployment time
- Minimize human errors
- Provides consistant configurations
- Allows manual changes
ANALOG vs DIGITAL
ANALOG is ONE active call per phone
DIGITAL is MULTIPLE active calls per phone broken into T1, E1 and ISDN standards
DS0
Quantization: 256 levels Sampling: 8,000 samples/second Coding: 8 bits/sample "Pulse Code Modulation" (PCM) 8,000 bytes per second 64,000 bits/second = 64 kb/s DS0 rate
There are three steps in voice digitization: quantization, sampling and coding.
The telephone system quantizes the voice signal to 256 levels. This number is chosen to reduce the quantization error, which would be heard as noise after the signal is reconstructed, so that a person can’t hear it on the line. The diagram shows bin numbers 127 and 128 around zero volts.
The second step is sampling. Since this is a voiceband signal, the frequency bandwidth is about 3000 Hz, and so the sampling rate must be at least 6001 times per second, following the Dr. Nyquist’s sampling theorem. To ensure that there are no aliasing errors, the telephone system samples more often: 8,000 samples per second.
The third step is coding. The telephone system uses 8 bits to code the value of each sample. This technique of using 8 bits per sample is called by some Pulse Code Modulation (PCM), which doesn’t really mean anything.
To determine the number of bits per second required, multiply the number of samples per second (8,000) by the number of bits per sample (8) to get 64,000 bits per second, or 64 kb/s for short.
This 64 kb/s rate is called a DS0 rate signal (Digital service level zero, or digital signal rate zero, just called “DS0s” in the business). This is the base rate of most transmission systems and digital voice systems. When someone talks about a channel on a digital transmission system, they usually mean a DS0.
Commercial transmission systems which are actually deployed were designed to carry digitized voice, and thus move multiple DS0s. Since they are digital systems, they can be easily be adapted to carry data or video as well as digitized voice.
CUCM - SCCP Phone registration
- SCCP phone obtains the Power (PoE or AC adapter).
- The phone loads its locally stored firmware image.
- The phone learns the Voice VLAN ID via CDP from the switch.
- The phone uses DHCP to learn its IP address, subnet mask, default gateway and TFTP server address.
- The phone contacts the TFTP server and requests its configuration file. Each phone has a customized configuration file named SEP.cnf.xml created by CUCM and uploaded to TFTP when the administrator creates or modifies the phone.
- The phone registers with the primary CUCM server listed in its configuration file. CUCM then sends the softkey template to the phone using SCCP messages.
Cisco SCCP IP Phone Startup Process
Okay, step one, the phone has to get power. Whether they get it from the switch or from that brick plugged into the wall. The phone is going to load a locally stored image at that point, and it’s also going to go on a little power-on self test too. So once we get power to it, it does a little power-on self test, gets that little locally stored image started, and now the switch provides VLAN information to the IP phone by the Cisco Discovery Protocol. So CDP kicks in here. The phone sends a DHCP request now, because it now knows what VLAN it’s a member of and it gets the IP information and option 150, the TFTP server address. Now the phone goes to the TFTP server and it now needs to download the configuration file. At that point within that configuration file it now knows a lot of information including the Communications Manager that it should register with. We can set up a group of Communications Manager in the graphical administrative interface that says, “you can try this Communications Manager, if that’s not available try the next and then finally try the third”, so you can list up to three. So now our Communications Manager sends the softkey template to the phone using the Skinny Client Control Protocol message.