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This chapter addresses the quality of service (QoS) requirements for implementations of IP telephone solutions over the enterprise WAN. By applying the prerequisite tools, you can achieve excellent quality voice, video, and data transmissions over an IP WAN, irrespective of media and even at low data rates.
This chapter includes the following sections:
The enterprise WAN model is shown in Figure 8-1.
Before voice and video can be placed on a network, it is necessary to ensure that adequate bandwidth exists for all required applications. To begin, the minimum bandwidth requirements for each major application (for example, voice, video, and data) should be summed. This sum represents the minimum bandwidth requirement for any given link, and it should consume no more than 75% of the total bandwidth available on that link. This 75% rule assumes that some bandwidth is required for overhead traffic such as routing and Layer 2 keepalives, as well as for additional applications such as e-mail and HTTP traffic. See Figure 8-2.
This section discusses the tools used to implement QoS for IP telephony applications over the enterprise WAN. These tools include traffic classification, prioritization, link fragmentation and interleaving (LFI), and traffic shaping. This section concludes with a summary of best practices for each of the applicable data link protocols.
Before traffic can be handled according to its unique requirements, it must be identified or labeled. There are numerous classification techniques for doing this. These include Layer 3 schemes such as IP precedence or the differentiated services code point (DSCP), Layer 2 schemes such as 802.1P, and implicit characteristics of the data itself, such as the traffic type using the Real-time Transport Protocol (RTP) and a defined port range.
In the majority of cases, traffic classification is done at the edge of the network by an Ethernet switch such as the Catalyst 6000. In these cases, the trust boundary is extended to the edge of the enterprise network and resides in the access or distribution layer. For a more detailed discussion of trust boundaries, see the "Trust Boundaries" section.
In some cases, however, the ability to classify and define a trust boundary at the edge of the network might not exist, such as in a small branch with an Ethernet switch that has no classification capabilities. In this situation, the trust boundary and classification can be achieved on the router itself. To facilitate this, Cisco recommends that the IP phones and PC have unique IP address ranges. For further details on IP addressing schemes, see the "IP Addressing and Management" section
In choosing from among the many available prioritization schemes, the major factors to consider include the type of traffic being put on the network and the wide area media to be traversed. For multiservice traffic over an IP WAN, Cisco recommends low-latency queuing for low-speed links. This allows up to 64 traffic classes with the ability to specify, for example, priority queuing behavior for voice and interactive video, a minimum bandwidth for Systems Network Architecture (SNA) data and market data feeds, and weighted fair queuing to other traffic types.
Figure 8-3 shows this prioritization scheme as follows:
The following points must be taken into account when configuring low-latency queuing:
Table 8-1 gives the minimum bandwidth requirements for voice, video, and data networks using Cisco CallManager Release 3.0. Note that these values are minimum, and any network should be engineered with adequate capacity.
| Traffic Type | Leased Lines | Frame Relay | ATM | ATM/Frame Relay |
|---|---|---|---|---|
Voice + data | 64 kbps | 64 kbps | 768 kbps | 768 kbps |
Voice, video, and data | 768 kbps | 768 kbps | 768 kbps | 768 kbps |
Because wide area bandwidth is often prohibitively expensive, only low-speed circuits may be available or cost effective when interconnecting remote sites. In these cases it is important to achieve the maximum savings by transmitting as many voice calls as possible over the low-speed link. Many compression schemes, such as G.729, can squeeze a 64-kbps call down to an 8- kbps payload. Cisco gateways and IP phones support a range of codecs that can enhance efficiency on these low-speed links.
The link efficiency can be further increased by using compressed RTP (cRTP), which compresses a 40-byte IP + UDP + RTP header to approximately two to four bytes. In addition, voice activity detection (VAD) takes advantage of the fact that, in most conversations, only a single party is talking at a time. VAD recovers this empty time and allows data to use the bandwidth.
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Note cRTP is currently supported only for leased lines and Frame Relay media. Cisco IOS Release 12.1(2)T, which greatly enhances performance, is the recommended system software for cRTP. |
For low-speed links (less than 768 kbps), it is necessary to use techniques that provide link fragmentation and interleaving (LFI). This places bounds on jitter by preventing voice traffic from being delayed behind large data frames. The two techniques that exist for this purpose are FRF.12 for Frame Relay and Multilink PPP for point-to-point serial links. Figure 8-4 depicts the general operation of LFI.
Traffic shaping is required for multiple access, non-broadcast media such as ATM and Frame Relay, where the physical access speed varies between two endpoints. Traffic shaping technology accommodates mismatched access speeds. In the case of Frame Relay with FRF.12, traffic shaping also allows delay variation, or jitter, to be bounded appropriately. For ATM, data rates are such that fragmentation is typically not required. Figure 8-5 demonstrates traffic shaping with Frame Relay and ATM.
Table 8-2 shows the minimum recommended software release for enterprise voice implemented over the WAN and includes recommended parameters for QoS tools. The currently recommended Cisco IOS versions will change with future releases.
| Data Link Type | Minimum Cisco IOS Release | Classification | Prioritization | LFI | Traffic Shaping |
|---|---|---|---|---|---|
Serial Lines | 12.0(7)T | IP prec = 5, DSCP = EF for voice; other classes of traffic have a unique classification | LLQ with CBWFQ | MLPPP | N/A |
Frame Relay | 12.1(2)T | IP prec = 5, DSCP = EF for voice; other classes of traffic have a unique classification | LLQ with CBWFQ | FRF.12 | Shape traffic to CIR |
ATM | 12.0(7)T | IP prec = 5, DSCP = EF for voice; other classes of traffic have a unique classification | LLQ with CBWFQ | Minimum bandwidth = 768 kbps | Shape traffic to guaranteed portion of bandwidth |
ATM/ | 12.1(2)T | IP prec = 5, DSCP = EF for voice; other classes of traffic have a unique classification | LLQ with CBWFQ | Minimum bandwidth = 768 kbps | Shape traffic to guaranteed portion of bandwidth |
Call admission control (CAC) is required to ensure that network resources are not oversubscribed. Calls that exceed the specified bandwidth are either rerouted using an alternative route such as the PSTN, or a busy tone is returned to the calling party. Figure 8-6 demonstrates that CAC is needed regardless of whether the implementation model is toll bypass or IP telephony to the desktop.
There are two schemes for providing CAC for voice calls over the WAN:
Posted: Thu Jul 27 10:34:11 PDT 2000
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