Enterprise QoS Solution Reference Network Design Guide

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How is QoS Optimally Deployed within the Enterprise? 1-14 Enterprise QoS Solution Reference Network Design Guide Chapter 1 Quality of Service Design Overview Note Higher delays may also be viewed as acceptable to certain organizations, but the corresponding reduction in VoIP quality must be taken into account when making such design decisions. Jitter buffers (also known as play-out buffers) are used to change asynchronous packet arrivals into a synchronous stream by turning variable network delays into constant delays at the destination end systems. The role of the jitter buffer is to balance the delay and the probability of interrupted playout due to late packets. Late or out-of-order packets are discarded. If the jitter buffer is set either arbitrarily large or arbitrarily small, then it imposes unnecessary constraints on the characteristics of the network. A jitter buffer set too large adds to the end-to-end delay, meaning that less delay budget is available for the network such that the network needs to support a delay target tighter than practically necessary. If a jitter buffer is too small to accommodate the network jitter, then buffer underflows or overflows can occur. An underflow is when the buffer is empty when the codec needs to play out a sample. An overflow is when the jitter buffer is already full and another packet arrives that cannot therefore be enqueued in the jitter buffer. Both jitter buffer underflows and overflows cause packets to be discarded. Adaptive jitter buffers aim to overcome these issues by dynamically tuning the jitter buffer size to the lowest acceptable value. Well-designed adaptive jitter buffer algorithms should not impose any unnecessary constraints on the network design by: Instantly increasing the jitter buffer size to the current measured jitter value following a jitter buffer overflow. Slowly decreasing the jitter buffer size when the measured jitter is less that the current jitter buffer size. Using a Programmable Logic Controller (PLC) to interpolate for the loss of a packet on a jitter buffer underflow. Where such adaptive jitter buffers are used, we can in theory engineer out explicit considerations of jitter by accounting for worst-case per hop delays. Advanced formulas can be used to arrive at network-specific design recommendations for jitter based on maximum and minimum per-hop delays. Alternatively, this 30 ms value can be used as a jitter target as extensive lab testing has shown that when jitter consistently exceeds 30 ms voice quality degrades significantly. Because of its strict service-level requirements, VoIP is well suited to the Expedited Forwarding Per-Hop Behavior, as defined in RFC 3246 (formerly RFC 2598). It should therefore be marked to DSCP EF (46) and assigned strict priority servicing at each node, regardless of whether such servicing is done in hardware (as in Catalyst switches via hardware priority queuing) or in software (as in Cisco IOS routers via LLQ). The bandwidth consumed by VoIP streams (in bps) is calculated by adding the VoIP sample payload (in bytes) to the 40-byte IP/UDP/RTP headers (assuming that cRTP is not in use), multiplying this value by 8 (to convert it to bits) and then multiplying again by the packetization rate (default of 50 packets per second). Table 1-2 details the bandwidth per VoIP flow at a default packet rate of 50 packets per second (pps). This does not include Layer 2 overhead and does not take into account any possible compression schemes, such as cRTP. Table 1-2 Voice Bandwidth (without Layer 2 Overhead) Voice Payload Packets per Bandwidth per Bandwidth Consumption Sampling Rate in Bytes Second Conversation G.711 20 ms 160 50 80 kbps Version 3.3
Chapter 1 Quality of Service Design Overview Call-Signaling Traffic Version 3.3 Table 1-2 Voice Bandwidth (without Layer 2 Overhead) How is QoS Optimally Deployed within the Enterprise? G.711 30 ms 240 33 74 kbps G.729A 20 ms 20 50 24 kbps G.729A 30 ms 30 33 19 kbps Note The Service Parameters menu in Cisco CallManager Administration can be used to adjust the packet rate. It is possible to configure the sampling rate above 30 ms, but this usually results in poor voice quality. A more accurate method for provisioning VoIP is to include the Layer 2 overhead, which includes preambles, headers, flags, cyclic redundancy checks (CRCs), and ATM cell-padding. The amount of overhead per VoIP call depends on the Layer 2 technology used: 802.1Q Ethernet adds (up to) 32 bytes of Layer 2 overhead. Point-to-point protocol (PPP) adds 12 bytes of Layer 2 overhead. Multilink PPP (MLP) adds 13 bytes of Layer 2 overhead. Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12 adds 8 bytes. ATM adds varying amounts of overhead, depending on the cell padding requirements. Table 1-3 shows a more accurate bandwidth provisioning example for voice because it includes Layer 2 overhead. Table 1-3 Voice Bandwidth (Including Layer 2 Overhead) Bandwidth Consumption 802.1Q Ethernet PPP MLP Frame-Relay w/FRF.12 ATM G.711 at 50 pps 93 kbps 84 kbps 86 kbps 84 kbps 106 kbps G.711 at 33 pps 83 kbps 77 kbps 78 kbps 77 kbps 84 kbps G.729A at 50 pps 37 kbps 28 kbps 30 kbps 28 kbps 43 kbps G.729A at 33 pps 27 kbps 21 kbps 22 kbps 21 kbps 28 kbps Note A handy tool for quickly and accurately calculating VoIP bandwidth requirements (factoring in the codec, the use of cRTP and L2 overhead) can be found at: http://tools.cisco.com/Support/VBC/jsp/Codec_Calc1.jsp The following are key QoS requirements and recommendations for Call-Signaling traffic: Call-Signaling traffic should be marked as DSCP CS3 per the QoS Baseline (during migration, it may also be marked the legacy value of DSCP AF31). 150 bps (plus Layer 2 overhead) per phone of guaranteed bandwidth is required for voice control traffic; more may be required, depending on the call signaling protocol(s) in use. Enterprise QoS Solution Reference Network Design Guide 1-15
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