Voice Quality Report Via Sip Publish - Cisco SPA301 Administration Manual

Getting Started
Ensuring Voice Quality
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide
dynamically adjust the size of the jitter buffer according to the network
conditions that exist during a call.
- The minimum jitter buffer size is 30 ms or 10 ms plus the current RTP
frame size, whichever is larger, for all jitter level settings. However, the
starting jitter buffer size value is larger for higher jitter levels. This setting
controls the rate at which the jitter buffer size is adjusted to reach the
minimum.
- Jitter Buffer Adjustment—Controls how the jitter buffer should be
adjusted.
• Echo—Impedance mismatch between the telephone and the IP telephony
gateway phone port can lead to near-end echo. Cisco SPA IP phones have
a near-end echo canceller with at least 8 ms tail length to compensate for
impedance mismatch. Cisco SPA IP phones implement an echo suppressor
with CNG so that any residual echo is not noticeable.
• Hardware noise—Certain levels of noise can be coupled into the
conversational audio signals because of the hardware design. The source
can be ambient noise or 60 Hz noise from the power adaptor. The Cisco
hardware design minimizes noise coupling.
• End-to-end delay—End-to-end delay does not affect voice quality directly,
but is an important factor in determining whether IP phone subscribers can
interact normally in a conversation. A reasonable delay should be 50 to
100 ms. End-to-end delay larger than 300 ms is unacceptable to most
callers. Cisco SPA IP phones support end-to-end delays well within
acceptable thresholds.
• Adjustable Audio Frames Per Packet—Allows you to set the number of
audio frames contained in one RTP packet. Packets can be adjusted to
contain from 1–10 audio frames. Increasing the number of frames
decreases the bandwidth utilized, but it also increases delay and can affect
voice quality.

Voice Quality Report via SIP Publish

SIP PUBLISH enables the collection and reporting of voice call quality information
derived from RTCP-XR and the call information from SIP is conveyed from a User
Agent in a session to the third party.
SIP PUBLISH is sent for each session terminating at the Reporter.
—Multiple Calls: When there are multiple calls (maximum 10) on SPA525G, SIP
PUBLISH is reported at the end of the session.
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