Term
| What are the different trunk types used with PBX? |
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Definition
- Interoffice truck - connects 2 CO switches (PSTN switch trunk)
- Tandem trunk - connects central offices with a geographical area
- Toll-connecting truck - connects the CO to the long distance office
- Intertoll truck - connects 2 long distance offices
- Tie Truck - connects 2 PBXs. Also called a private trunk
- PBX-to-CO trunk - connects the CO switch to the enterprise PBX |
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Term
| What are some of the different ports that can be used to connect voice end stations and private voice switches? |
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Definition
- Foreign exchange station
- Foreign exchange office
- Ear and mount (E&M)
- Channelized T1 (or E1)
- ISDN primary rate interface (PRI) |
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Term
| What are the steps involved in converting analog to digital? |
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Definition
1. Filtering
2. Sampling
3. Digitizing |
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Term
| What is skinny station control protocol used for in VoIP? |
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Definition
| SSCP is a client/server signaling protocol for call setup and control using TCP. IP phones use SSCP to register with the call manager and to establish calls. SSCP is used for VoIP call signaling and for features such as message waiting indicators. |
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Term
| Briefly describe PBX switches |
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Definition
| Traditional switches and PBX route voice use TDM technology and use 64kbps circuits. It uses a private network and proprietary protocols. PBX are customer owned voice switches and may scale up to 1000 phones. Calls that are placed between offices go through the private voice network (on-net) and calls outside the private network are routed top the local PSTN (off-net) |
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Term
| What is the use of DHCP, DNS, and TFTP in VoIP? |
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Definition
| IP phones use DHCP to obtain their IP addressing information. TFTP is used to download the IP phone's operating system and configuration. DNS can be used to find the servers |
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Term
| Describe the IPT multisite distributed WAN call-processing model |
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Definition
| The multisite distributed WAN call-processing model is a solution for large enterprises with several large locations. Server CM clusters are deployed at the large sites for redundancy and unity server are used for messaging. This model also supports remote sites to be distributed off the large sites. |
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Term
| Describe how LLQ works with QoS in a VoIP network |
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Definition
| LLQ provides a single priority queue, like PQ-FWQ, but its preferred for VoIP networks because it can also be configured for guarantee bandwidth for different classes or traffic. |
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Term
| How do real time protocol (RTP) and real time control protocol (RTCP) work with regards to VoIP? |
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Definition
In VoIP, RTP transports audio streams. Its a transport layer protocol that carries digitized voice in its payload, using UDP.
RTCP is a session layer protocol that monitors the delivery of data, and provides control and identification functions. RTP runs on an even port, and RTCP runs on the next odd port. |
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Term
| How does media gateway control protocol work with regards to VoIP? |
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Definition
| MGCP is a client/server signaling protocol used to control gateways in VoIP networks. Its primary function is to control and supervise connection attempts between different media gateways. MGCP gateways handle translation between audio signals and the IP network. |
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Term
| How is voice activity detection used with VoIP networks? |
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Definition
| Using VD, you can suppress packets of silence. Silence suppression at the source IP telephone or VoIP gateway increases the calls or data volumes that can be carried over the links, more effectively utilizing network bandwidth. Bandwidth savings are at least 35% in conservative estimates. |
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Term
| Describe how auto QoS works with QoS in VoIP networks |
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Definition
| Auto QoS uses a simpler CLI to enable QoS for VoIP in WAN and LAN environments. It significantly reduces the amount of configuration lines necessary to support VoIP in the network |
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Term
| Describe how PQ-FWQ is used for QoS in VoIP networks |
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Definition
| PQ-WFQ adds a single priority queue to WFQ. The priority queue is used for VoIP packets. With PQ-WFQ, the router places VoIP RTP packets in a strict priority queue that is always serviced first |
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Term
| Describe the single-site IPT deployment model |
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Definition
| The single-site deployment model is a solution for enterprise located in a single large building or campus area with no voice on the WAN links and no remote sites. |
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Term
| What are the typical bandwidth requirements when designing a VoIP network? |
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Definition
- G.729 calls use 26kpbs
- G.711 calls use 80kpbs
The total bandwidth for voice, data, and video should not exceed 75% sustained of the provisioned link capacity during peak times. G.729 is the recommended codec for calls over the WAN because of its lower bandwidth requirements. |
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Term
| What signaling categories need to occur for a call to be placed, managed, and closed? |
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Definition
- Supervisory provides call control and phone state (on-hook and off-hook)
- Addressing provides dialed digits
- Informational provides information such as dial and busy tunes, and progress indicators |
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Term
| What are the 2 major forms of digital signaling? |
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Definition
- Channel associated signaling (CAS) - the signaling is included in the same channel as the voice call
- Common channel signaling (CCS) - the signaling is provides in a separate channel |
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Term
| Describe the unified call manage express deployment model |
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Definition
| Call manage express (CMX), unity express and IPCC express provide the same services on the router. Its a low cost solution for small branch offices. |
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Term
| What are the different delay components involved with VoIP? |
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Definition
- Propagation delay - How long it takes a packet to travel between 2 endpoints
- Processing delay - includes coding, compression, decoding and decompression delay
- Serialization delay - how long it takes to place the bits on the circuit
- Dejitter delay - caused because packets can take different, redundant paths to reach the destination. So packets may not arrive at the constant rate because they take different paths. |
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Term
| Describe the IPT multisite centralized WAN call-processing model |
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Definition
| The centralized WAN call-processing model is a solution for medium enterprises with one large location and many remote sites. A CM cluster with multiple servers is deployed for redundancy at the large site. Remote site IP phones register to the CM cluster located in the main site. Remote sites use voice-enabled gateway routers with SRST for redundancy. On the WAN, QoS features are configured to prioritize the VoIP packets over the other packet types. |
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Term
| Describe now LFI is used for QoS in VoIP networks |
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Definition
| LFI is used to reduce the serialization delay in a multi-service network. Small VoIP packets have to compete with large data traffic packets for outbound interfaces. When the large packet is fragmented into smaller packets, the VoIP packets can be interleaved between the data packets. |
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Term
| Describe the use of session initiation protocol with regards to VoIP |
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Definition
| SIP is an application layer control (signaling) protocol for creating, modifying, and terminating internet multimedia conferences, internet phone calls, and multimedia distributions. |
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Term
| What are the 4 major functional areas in Cisco's IPT architecture? |
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Definition
- Client endpoints - includes the IP phones, analog and digital gateways and digital signal processor (DSP) farms
- Call processing - using Cisco's unified call manage (CM), the CM servers are the "brains" of the voice dial plan and are used to establish IPT called between IP phones.
- Service applications - includes IVR, auto attendant, and unity unified messaging system for voice mail. Cisco IP contact center (IPCC) is used for enterprise call center applications
- -Voice enabled infrastructure - includes QoS-enabled devices such as LAN switches and routers. These devices are configured to be IPT-aware and provide secure guarantees to the VoIP traffic |
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Term
| How does CRTP help QoS in VoIP networks? |
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Definition
| CRTP compresses IP/UDP/RTP headers from 40 bytes to 2-4 bytes. It is recommended for links lower than 768kpbs, but should not be configured on a router with CPU utilization above 75% |
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Term
| The variable delay of received VoIP packets is corrected with what kind of buffer? |
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Definition
| Dejitter buffers are used at the receiving end to smooth out the variable delay of received packets. |
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Term
| An organization uses what type of system to route calls to agents based on the agent skill group or call statistics? |
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Definition
| Automatic Call distribution (ACD) system. ACD are used by airline reservation systems, customer service departments , and other call centers. |
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Term
| What is the difference between a fixed delay and a variable delay? |
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Definition
| Fixed-delay components include processing, serialization, dejitter, and propagation delays. Variable-delay components include only queuing delays. |
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Term
| What is the recommended maximum one-way delay for G.114 for acceptable voice? |
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Definition
| 120-ms one-way maximum delay |
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Term
| What is the recommended max RTT for implementing an IPT solution? |
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Definition
| The RRT is 300ms. The one-way delay should not be more than 150ms |
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Term
| Which standard establishes specifications for call setup and packet format for VoFR? |
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Definition
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Term
| What protocol is preferred for inter-PBX trunks? |
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Definition
| Q.SIG is the preferred protocol for inter-PBX trunks |
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Term
| Digitizing is divided into what 2 processes? |
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Definition
| The digitizing process is divided into companding, and quantization and coding. |
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Term
| What is the purpose of Call Admission Control (CAC) |
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Definition
| Call Admission Control (CAC) prevents new voice calls from affecting existing voice calls. |
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