© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Identifying Voice Networking Considerations Identifying the Requirements of Voice Technologies

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Voice Quality Considerations Examine the possible causes of packet loss and delay in the initial design. Use QoS mechanisms as a groundwork for a high-quality voice network.

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Fixed Network Delay Considerations Sources of delay: Propagation delay: 6 ms per km Serialization delay: frame length / bit rate Processing delay: depends on codec –Coding and compression –Packetization Solutions: None Faster link, smaller packets Hardware DSPs, coding algorithm

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Variable Network Delay Considerations Sources of delay: Queuing delay (variable packet sizes and number of packets) Dejitter buffers Solutions: Link fragmentation and interleaving Constant delay, uncongested network

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Jitter Variation in the delay of received packets Caused by network congestion, improper queuing, or configuration errors

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Packet Loss Causes voice clipping Caused by: –Congested links –Improper network QoS configuration –Bad packet buffer management on the routers –Routing problems Up to 30 ms of lost voice correctable by DSP using interpolation Packet losses up to one packet correctable with no voice quality degradation

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Problem of Echo

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Echo Cancellers Reduce the Level of Echo

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Voice Coding and Compression The quality of transmitted speech is a subjective listener response. MOS is a common benchmark to define sound quality. MOS scales from 1 (bad) to 5 (excellent). ITU StandardData Rate*MOS Score PCM G kbps4.1 ADPCM G.726/G /24/32/40 kbps 3.85 or less LD-CELP G kbps3.61 CS-ACELP G.7298 kbps3.92 ACELP/MPMLQ G /5.3 kbps3.9/3.65 *Note: Data rates shown are for digitized speech only and do not include overhead of RTP, UDP, IP, and Layer 2 headers.

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Example: Codec Complexity and Calls per DSP on the Cisco AS54-PVDM2-64 Module Low Complexity (Maximum 64 Calls) Medium Complexity (Maximum 32 Calls) High Complexity (Maximum 24 Calls) G.711 a-lawG.729aG.723.1: 5.3K and 6.3K G.711 mu-lawG.729abG.723.1A: 5.3K and 6.3K Fax passthroughG.726: 16K, 24K, and 32KG.728 Modem passthroughT.38 fax relayModem relay Clear-channel codecCisco Fax RelayAMR-NB: 75K, 5.15K, 5.9K, 6.7K, 7.4K, 7.95K, 10.2K, 12.2K, and silence insertion descriptor

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Bandwidth Availability Goal: Reduce the amount of traffic per voice call Solutions: –Use an effective voice coding and compression mechanism. –Compress IP headers by using compressed Real-Time Transport Protocol. –Suppress packets of silence by using voice activity detection.

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Calculating Voice Bandwidth Voice packet size = (Layer 2 header) + (IP/UDP/RTP header) + voice payload Voice packets per second (pps) = (codec bit rate) / (voice payload size) Bandwidth = (voice packet size) * (pps) Example for G.729 call with 8-kbps codec bit rate with cRTP and 20 bytes voice payload: –Voice packet size = 6 bytes + 2 bytes + 20 bytes = 28 bytes –Voice packet size = 28 bytes * 8 bits/byte = 244 bits –Voice pps = 8000 bits/sec / 160 bits/packet = 50 pps –Bandwidth = 244 bits * 50 pps = 11.2 kbps

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Example: Voice Codec Bandwidth Calculator for G.729 Codec

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Voice Bandwidth and Codec Standards CompressionPayload Size Bandwidth with cRTP No. of Calls on a 512-kbps Link (without cRTP/ with cRTP) G.711 (64 kbps) /7 G.726 (32 kbps) /14 G.726 (24 kbps) /17 G.728 (16 kbps) /26 G.729 (8 kbps) /46 G (6.3 kbps) /64 G (5.3 kbps) /73

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Enterprise QoS Mechanisms for Voice Traffic classification Queuing or scheduling Bandwidth provisioning and call admission control

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Access Layer QoS Mechanisms for Voice 802.1Q trunking and 802.1p Multiple egress queues Traffic classification and network trust boundary Layer 3 awareness and the ability to implement QoS access control lists

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Recommended Practice: Separate Voice and Data VLANs Voice device protection from external networks QoS trust boundary extension to voice devices Protection from malicious network attacks Ease of management and configuration

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Example: QoS Networking Mechanisms

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Example: Low Latency Queuing

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v QoS Consideration for Voice in the WAN WAN QoS mechanisms: Bandwidth provisioning Traffic classification Queuing and scheduling Traffic shaping Link efficiency techniques Call admission control

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Call Admission Control Protects voice traffic from being negatively affected by other voice traffic Keeps excess voice traffic off the network Reroutes excess voice traffic in the following scenarios: –Call rerouted via an alternate packet network path –Call rerouted via the PSTN network path –Call returned to the originating TDM switch with the reject cause code

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Example: Call Admission Control VoIP Network Without CAC VoIP Network with CAC

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Implementing CAC with RSVP RSVP is an industry-standard signaling protocol that enables an application to reserve bandwidth dynamically. RSVP signaling messages are exchanged between the source and destination devices. RSVP process interacts with the QoS manager on router interfaces to "reserve" bandwidth resources. Calls are admitted or rejected based on the outcome of the RSVP reservations.

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Traffic Engineering Terms Grade of service Erlang Centum call seconds Busy hour Busy hour traffic Blocking probability Call Detail Record

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Erlang Tables Show erlangs of offered traffic, number of circuits, and grade of service Three common erlang tables: –Erlang B assumes that calls receiving a busy signal are immediately cleared. –Extended Erlang B assumes that a certain percentage of calls receiving a busy signal are redialed. –Erlang C assumes that blocked calls are queued.

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Example: Erlang B Table Blocking Probability Number of Circuits Number of erlangs decreases with the decreased blocking probability. Number of erlangs increases with the number of simultaneous connections. Busy hour traffic (BHT) in erlangs

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v Summary Voice quality in an IP network is directly affected by delay, jitter, and packet loss. An echo is the audible leak of the voice of the caller into the receive (return) path. Voice communication over IP relies on voice that is coded and encapsulated into IP packets. A primary WAN issue when network designers are designing voice on IP networks is bandwidth availability. QoS mechanisms are important for networks that carry voice. Traffic engineering is a science of selecting the right number of lines and the proper types of service to accommodate users.

© 2007 Cisco Systems, Inc. All rights reserved.DESGN v