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Each site in a multisite deployment usually is interconnected by an IP WAN, or occasionally
by a metropolitan-area network (MAN) such as Metro Ethernet. Bandwidth on WAN links
is limited and relatively expensive. The goal is to use the available bandwidth as efficiently
as possible. Unnecessary traffic should be removed from the IP WAN links through content
filtering, firewalls, and access control lists (ACL). IP WAN acceleration methods for
bandwidth optimization should be considered as well. Any period of congestion could
result in service degradation unless QoS is deployed throughout the network.
Voice streams are constant and predictable for Cisco audio packets. Typically, the G.729
codec is used across the WAN to best use bandwidth. As a comparison, the G.711 audio
codec requires 64 kbps, whereas packetizing the G.711 voice sample in an IP/UDP/RTP
header every 20 ms requires 16 kbps plus the Layer 2 header overhead.
Voice is sampled every 20 ms, resulting in 50 packets per second (pps). The IP header is
20 bytes, whereas the UDP header is 8 bytes, and the RTP header is 12 bytes. The 40 bytes
of header information must be converted to bits to figure out the packet rate of the overhead.
Because a byte has 8 bits, 40 bytes * 8 bits in a byte = 320 bits. The 320 bits are sent
50 times per second based on the 20-ms rate (1 millisecond is 1/1000 of a second, and
20/1000 = .02). So:
.02 * 50 = 1 second
320 bits * 50 = 16,000 bits/sec, or 16 kbps
Bandwidth Challenges 7
NOTE This calculation does not take Layer 2 encapsulation into consideration. You can
find more information by reading the QoS Solution Reference Network Design (SRND)
(http://www.cisco.com/go/srnd) or Cisco QOS Exam Certification Guide, Second
Edition (Cisco Press, 2004).
Voice packets are benign compared to the bandwidth consumed by data applications. Data
applications can fill the entire maximum transmission unit (MTU) of an Ethernet frame
(1518 bytes or 9216 bytes if jumbo Ethernet frames have been enabled). In comparison to
data application packets, voice packets are very small (60 bytes for G.729 and 200 bytes for
G.711 with the default 20-ms sampling rate).
In Figure 1-2, a conference bridge has been deployed at the main site. No conference bridge
exists at the remote site. If three IP Phones at a remote site join a conference, their RTP
streams are sent across the WAN to the conference bridge. The conference bridge, whether
using software or hardware resources, mixes the received audio streams and then sends
back three unique unicast audio streams to the IP Phones over the IP WAN. The conference
bridge removes the receiver's voice from his or her unique RTP stream so that the user does
not experience echo because of the delay of traversing the WAN link and mixing RTP audio
streams in the conference bridge.
Figure 1-2 Resource Challenges
Centralized conference resources cause bandwidth, delay, and capacity challenges in the
voice network. Each G.711 RTP stream requires 80 kbps (plus the Layer 2 overhead),
resulting in 240 kbps of IP WAN bandwidth consumption by this voice conference. If the
conference bridge were not located on the other side of the IP WAN, this traffic would not
8 Chapter 1: Identifying Issues in a Multisite Deployment
need to traverse the WAN link, resulting in less delay and bandwidth consumption. If the
remote site had a CUCM region configuration that resulted in calls with the G.729 codec
back to the main site, the software conferencing resources of CUCM would not be able to
mix the audio conversations. Hardware conferencing or hardware transcoder media resources
in a voice gateway are required to accommodate G.729 audio conferencing. Local hardware
conference resources would remove this need. All centrally located media resources (Music
On Hold [MOH], annunciator, conference bridges, videoconferencing, and media termination
points) suffer similar bandwidth, delay, and resource exhaustion challenges.