Cisco Systems 8961 Manual

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Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 8.5 (SIP)

OL-20861-01

Chapter 6 Understanding the VoIP Wireless Network

Components of the VoIP Wireless Network

Beginning with Cisco IOS release 12.2(11)JA, Cisco Aironet APs support the contention-based channel

access mechanism called Enhanced Distributed Coordination Function (EDCF). The EDCF-type of QoS

has up to eight queues for downstream (toward the 802.11b/g clients) QoS. You can allocate the queues

based on these options:

• QoS or Differentiated Services Code Point (DSCP) settings for the packets

• Layer 2 or Layer 3 access lists

• VLANs for specific traffic

• Dynamic registration of devices

Although you can have up to eight queues on the AP, you should use only two queues for voice traffic

to ensure the best possible voice QoS. Place voice (RTP) and signaling (SCCP) traffic in the highest

priority queue, and place data traffic in a best-effort queue.Although 802.11b/g EDCF does not

guarantee that voice traffic is protected from data traffic, you should get the best statistical results by

using this queuing model.

Note The Cisco Unified IP Phone marks the SCCP signaling packets with a DSCP value of 24 (CS3)

and RTP packets with DSCP value of 46 (EF).

To improve reliability of voice transmissions in a nondeterministic environment, the Cisco Unified IP

Phone supports the IEEE 802.11e industry standard and is Wi-Fi Multimedia (WMM) capable. WMM

enables differentiated services for voice, video, best effort data and other traffic. However, in order for

these differentiated services to provide sufficient QoS for voice packets, only a certain amount of voice

bandwidth can be serviced or admitted on a channel at one time. If the network can handle “N” voice

calls with reserved bandwidth, when the amount of voice traffic is increased beyond this limit (to N+1

calls), the quality of all calls suffers.

To help address the problems of VoIP stability and roaming, an initial Call Admission Control (CAC)

scheme is required. With CAC, QoS is maintained in a network overload scenario by ensuring that the

number of active voice calls does not exceed the configured limits on the AP. The Cisco Unified IP

Phone can integrate layer 2 TSpec admission control with layer 3 Cisco Unified Communications

Manager admission control (RSVP). During times of network congestion, calling or called parties

receive a fast busy indication. The system maintains a small bandwidth reserve so wireless phone clients

can roam into a neighboring AP (AP), even when the AP is at “full capacity.” After reaching the voice

bandwidth limit, the next call is load-balanced to a neighboring AP without affecting the quality of the

existing calls on the channel.

Implementing QoS in the connected Ethernet switch is highly desirable to maintain good voice quality.

The COS and DSCP values that the Cisco Unified IP Phone sets do not need to be modified.

The DSCP, COS and UP (WMM) markings correctly for the optimum transmission of video frames.

Note The Cisco Unified IP Phone 9971 does not support Video CAC; however, Voice CAC is supported for

WLANs.