Cisco AS5350XM Configuration Manual page 170

Enabling QoS Features for VoIP
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Tandem switching (also called dual encodings or dual compressions) can cause additional problems.
Note
Digital calls routed to a tandem (toll) office are converted there to analog form for processing, and then
reconverted to digital form for further transmission. Converting and reconverting in this way more than
about twice distorts signals irreparably. If your calls are subject to significant toll-office processing,
choose PCM if you have sufficient bandwidth. We also recommend that you employ a Cisco IOS
Multimedia Conference Manager (H.323 gatekeeper) or management application such as Cisco Voice
Manager to help manage these types of processes.
Other factors that might enter into your decision, or that you can use to tweak performance, include the
likelihood of multiple tandem encodings and how you handle packet fragmentation.
Tip For more information and configuration options, see the VoIP over PPP Links with Quality of Service
(LLQ / IP RTP Priority, LFI, cRTP) document, available online at
http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html
RTP Packet-Header Compression
Because of the repetitive nature of subsequent IP/UDP/RTP (network/transport/session-layer) headers,
you can compress them significantly. A recommended methodology is cRTP (Compressed Real-Time
Transfer Protocol), which, by tracking first-order and second-order differences between headers on
subsequent packets, compresses the 40-byte header to just 2 or 4 (without or with UDP checksum) bytes.
Other methodologies may be preferable if the cRTP high CPU usage causes delay. Use a compression
methodology on both ends of low-bandwidth (< 1.5 Mbps) WAN circuits, but not at all on high-speed (>
1.5 Mbps) WANs.
Tip For more information and configuration options, see the VoIP over PPP Links with Quality of Service
(LLQ / IP RTP Priority, LFI, cRTP) document, available online at
http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html
Serialization Delay
You can control packet (payload) size—which, in turn, controls how long one packet takes to be placed
on the system interface. Set this in bytes, ideally equaling no more than 20 ms (typically equivalent to
two 10-ms voice samples per packet). Increasing serialization delay increases end-to-end delay. You
want to incur no more than 150–200 ms of one-way, end-to-end delay.
Note Take care when you assign a payload size for your chosen codec. To assign a codec and payload size,
you use the codec codec bytes payload_size command under the dial-peer voip command. Although the
codec command permits a wide range of payload sizes, the universal port and voice feature cards permit
a much smaller range of sizes, to help ensure that end-to-end delay for voice signals does not exceed
200 ms. If your network uses a variety of gateway and router types, you may need to ensure that payload
sizes are set both optimally (so as not to incur excessive end-to-end delay) and consistently.
Cisco AS5350XM and Cisco AS5400XM Universal Gateways Software Configuration Guide
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Configuring Voice over IP