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Design Guide for Cisco Unity Release 5.x
OL-14619-01
Chapter 6 Integrating Cisco Unity with the Phone System
Integrating with Cisco Unified Communications Manager Express (by Using SCCP or SIP)
Packetization (SCCP Integrations Only)
The Real-Time Transport Protocol (RTP) is used to send and receive audio packets over the IP network.
Each discrete packet has a fixed-size header, but the packets themselves can vary in size, depending on
the size of the audio stream to be transported (which varies by codec) and the packetization setting. This
variable size function helps utilize network bandwidth more efficiently. Reducing the number of packets
that are created per call sends fewer total bytes over the network.
Packetization is set in the Cisco Unified CM Service Parameters, in the Preferred G711 Millisecond
PacketSize and Preferred G729 Millisecond PacketSize parameters. Cisco
Unity supports any packet
size up to 30ms for G.711 audio, and any packet size up to 60 ms for G.729a audio. The default setting
is 20ms for both; there may be latency issues with lower settings.
DSCP is a priority setting on each packet. DSCP helps intermediary routers manage network congestion
and lets them know which packets to prioritize ahead of others. Following Cisco AVVID standards, the
Cisco
Unity-CM TSP marks the SCCP packets (call control) with a default DSCP value of 26 (the TOS
octet is 0x68), and the RTP packets (audio traffic) with a default DSCP value of 46 (the TOS octet is
0xB8). Thus, the RTP audio packets can be assigned priority over other packets by using the router
settings. Note that even though Cisco Unified
CM allows you set different DSCP values, when integrated
with Cisco
Unity, the DSCP values set by the Cisco Unity-CM TSP always take precedence.
With each new audio stream (once per call), Cisco Unified CM tells Cisco Unity which packet size to
use, and the Cisco
Unity-CM TSP sets the DSCP priority for the stream. The entire stream (call) stays
at the specified packet size and priority. For example, an audio stream could be broken up into packets
of 30ms each. A 30ms G.729a audio stream would be 30 bytes plus the header per packet, and a 30ms
G.711 stream would be 240 bytes plus the header per packet. For information on setting Cisco
Unified
CM Service Parameters, see the Cisco Unified CM documentation at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_series_home.html.
Integrating with Cisco Unified Communications Manager
Express (by Using SCCP or SIP)
Cisco Unity supports Cisco Unified Communications Manager (CM) Express (formerly known as Cisco
Unified CallManager Express) integrations through both SCCP and SIP interfaces.
Figure 6-8 shows the
connections.
Figure 6-8 Cisco Unity SCCP and SIP Connections to Cisco Unified Communications Manager
Express Over a LAN
See Table 6-5 for information on the differences in these integration methods.
Cisco
Unity server
Cisco Unified
Communications
Manager
Express
LANPSTN
191850