About this Blog
In this blog I will cover the basic Interview question related to Cisco IP telephony.
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Q. What is CUCM ( Cisco Call Manager )?
It is web based application that allow us to make individual and manual configuration changes to the CUCM Nodes. It serves as software based call processing component of Cisco unified Communication family of products.
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Q. What is CUCM Function?
1) Call Processing
2) Signalling & Device Control
3) Dial Plan Administration
4) Phone feature Administration
5) Directory Services
6) OMA&P ( Operation, maintenance, administration and programming)
7) Programming interface to external voice processing application ( Ex. CIPC ( Softphone))
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Q. What is CUCM Cluster?
It is a set or group of networked server that can be used to provide specific services per server.
CUCM Cluster may contain 20 server out of which a maximum of 8 call processing "subscriber" are allowed, the other server nodes within the cluster may be configured as dedicated Publisher, dedicated TFTP Subscriber, Music On Hold Server or Media resource subscriber.
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Q. What is CUCM TFTP Server ?
- It is a critical service for IP Phone.
- IP Phone uses TFTP Server to download its configuration file ( File name : SEP < Macaddress> cnf.xml) also the phone download the firmware and other data from TFTP Server.
- Without TFTP the phone simply do not function.
Q. Difference Between Publisher and Subscriber ?
- "Publisher" refer to the server that is the first node in the cluster and contain writable copy of configuration database.
It is used for administration purpose and possibly as a fallback for call processing on small cluster.
- "Subscriber" are the second and all other nodes which just contain a replicated copy of the Database.
It is used for performing active call processing duties.
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Q. Explain Cisco IP Phone Boot up Process step by step ?
1) Cisco IP phone connect to an Ethernet Switch port, If the IP phone and Switch Support PoE, the IP phone receive power through Cisco Proprietary PoE or 802.3af Standard.
[On phone :"We can see Mute, handset and Speaker buttons are illuminated"]
2) Then Phone Load its locally stored firmware image, After that Cisco switch will deliver Voice Vlan Information to the IP Phone Using CDP( Cisco Discovery Protocol )as a delivery mechanism.
[Phone displaying : "Configuring Vlan"]
3) IP Phone send DHCP request asking for an IP address on its Voice Vlan
[Phone displaying : "Configuring IP"]
DHCP server respond with IP address, Subnet Mask, Default Gateway, DNS and Domain name information and TFTP server IP address through option 150.
4) Now Cisco phone has IP address of TFTP Server, the phone contact the TFTP server and request its configuration file.
5) Each phone has customized configuration file named -SEP < Macaddress> cnf.xml created by CUCM and uploaded to TFTP server when the administrator creates or modifies the phone.
6) The phone register with primary CUCM server listed in its configuration file. the cucm send the Directory number & Softey Template to the using SCCP Messages.
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Q. What is ANI and DNIS?
ANI stands for Automatic Number Identification also used to represent "Calling Party Number".
DNIS stands for Dialed Number Identification System also used to represent "Called Party Number"
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Q. What is OnNet and OffNet Calls ?
OnNet call is one where both the called and calling parties are on the same network, Unlike
OffNet call where the called and calling parties are on different network.
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Q. Difference between Translation Pattern vs Transformation Pattern ?
Translation Pattern : Translation Pattern are used for modifying the calling / called party number based on dialed number, This is done before a destination is selected.
It means calls route through translation pattern to modify the calling / called party number before reaching a route pattern which points to gateway or end device such as a phone.
Transformation Pattern : Transformation Pattern are used for modifying the calling / called party number once an endpoint or destination has been selected.
Transformation pattern are used to change the calling / called party number right before a call is sent out of a gateway or trunk or used to modify the displayed number on a phone when a call is received.
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Q. What is SRST & MGCP Fallback? Explain with examples.
SRST : means "Survivable remote site telephony" it is basically enabling the capability of router to register phone with itself in case of WAN failure in a multi-site deployment model with centralized call processing.
MGCP Fallback : It is used for remote voice gateway connecting to PSTN, So that in case they lose connectivity to CUCM they could fallback to default services i.e H.323, thus the link to PSTN still remain up.
Examples :
SRST Example:
Lets understand with above figure, Site B Phone are coming over the wan and register with CUCM located at Site A.So for redundancy purpose what we need is that in case of WAN failure the Site B phones should get register on the Site B gateway ( When Wan link is down ), that is Site B gateway configured as SRST.
MGCP Fallback Example:
Now in above figure let say if Site B MGCP gateway loose connectivity with CUCM the gateway fallback to default service i.e H.323 and link with PSTN remains up.
If Site B gateway is a H.323 type, then you don't need MGCP fallback configuration at all it is only required if Site B gateway configured as MGCP.
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Q. What is device pool ?
1) It define set of common characteristics for device
2) It contain system, device and location related information.
3) The common device configuration window records all user oriented information.
4) After creating device pool we can use it configure devices such as Cisco IP Phones, Gateway, Conference bridge, Transcoder, Media Termination Point, Voice Mail Ports and CTI Route Point.
5) Before Configure a device pool we must configure following items.
- CUCM
- Region & Date/ Time Group
- Softkey Template
- Optional : SRST Reference, Media resource group list, CSS for Auto registration, Location, AAR Group, AAR CSS.
Q. What is difference b/w SIP Early Media & Delay Media ?
Early media - The media b/w the Callee and Calling party established before the session established b/w the two, Also Callee party sends a message 183 instead of 180 and let the calling party establish a bearer channel b/w the 2 party so that they can share the message directly.
Delay media - In delay media the media is established after the sip session negotiation is completed.
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Q. Difference b/w SIP Early Offer Vs Delay Offer?
Early Offer- SIP uses SDP( Session Description Protocol ) in order to Negotiate Capability b/w two Parties. In Early offer SDP message is sent from the calling party to callee party in the initial invite message and the callee party respond with the negotiated capability in the 200OK it send to the calling party.
A---------------------------B
------INVITE(SDP)------>
<---200OKAY(SDP)------
------------RTP--------------
------------ACK----------->
Delay Offer - In delay offer Called party send the SDP message in the 200Ok message to the calling party and the calling party respond back with the negotiated parameter in the ACK message it send to the callee party.
A-------------------------B
---------INVITE--------->
<-------200OK(SDP)----
---------ACK(SDP)------>
--------------RTP------------
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Q. What is Extension-Mobility?
Extension-Mobility allow Users to access their Phone profile across location within Organisation, After Login into Cisco phone using Extension-Mobility feature once can access options like DN, Call history, Speed dials and Voicemails.
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Q. List of CUCM Certificates extensions ( CSR )
Call Manager ECDSA
CAPF
TVS
Tomcat ECDSA
IPsec
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Q. Difference b/w Restricted and Unrestricted version of Cisco Call Manager (CUCM)
Before discussing about difference, please keep this thing in mind. Unrestricted version of CUCM cannot be converted to Restricted version of CUCM once it is installed.
Difference : Unrestricted Version of CUCM does not provide signalling and media Encryption.
Restricted Version of CUCM has Encryption enabled, Whereas Unrestricted of CUCM does not have Encryption Enabled.
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