VoIP enables Cisco routers and access servers to carry voice traffic
(for example, telephone calls and faxes) over an IP network. In VoIP, the
digital signal processor (DSP) segments the voice signal into frames that
are then coupled in groups of two and stored in voice packets. These voice
packets are transported using IP in compliance with ITU-T specification
H.323. Because VoIP is a delay-sensitive application, you need to have a
well-engineered network end-to-end to successfully use VoIP. Fine-tuning
your network to adequately support VoIP involves a series of protocols and
features geared toward QoS. Traffic shaping considerations must be taken
into account to ensure the reliability of the voice connection.
VoFR enables a Cisco device to carry voice traffic (for example,
telephone calls and faxes) over a Frame Relay network. When voice traffic is
sent over Frame Relay, the voice traffic is segmented and encapsulated for
transit across the Frame Relay network. The segmentation engine uses FRF.12
fragmentation. FRF.12 (also known as FRF.11 Annex C) allows long data frames
to be fragmented into smaller pieces and interleaved with real-time frames.
In this way, real-time voice and nonreal-time data frames can be carried
together on lower speed links without causing excessive delay to the
real-time traffic.
The segmentation size configured must match the line rate, or the port
access rate. To ensure a stable voice connection, you must configure the
same data segmentation size on both sides of the voice connection. When
voice segmentation is configured, all priority queueing, custom queueing,
and weighted fair queueing is disabled on the interface.
When you configure voice and data traffic over the same Frame Relay DLCI,
you must take traffic shaping considerations into account to ensure the
reliability of the voice connection.
Cisco VoFR implementation supports the following types of VoFR calls:
- Static FRF.11 trunks
- Switched VoFR calls:
- Dynamic switched calls
- Cisco-trunk (private line) calls
VoATM enables a Cisco MC3810 multiservice concentrator to carry voice
traffic (for example, telephone calls and faxes) over an ATM network. The
Cisco MC3810 multiservice concentrator supports compressed VoATM on
ATM port 0 only.
When voice traffic is sent over ATM, the voice traffic is encapsulated
using a special AAL5 encapsulation for multiplexed voice. The ATM permanent
virtual circuit (PVC) must be configured to support real-time voice traffic,
and the AAL5 voice encapsulation must be assigned to the PVC. The PVC must
also be configured to support variable bit rate (VBR) for real-time networks
for traffic shaping between voice and data PVCs.
Traffic shaping is necessary so that the carrier does not discard the
incoming calls from the MC3810. To configure voice and data traffic shaping,
you must configure the peak, average, and burst options for voice traffic.
Configure the burst value if the PVC will be carrying bursty traffic. The
peak, average, and burst values are needed so the PVC can effectively handle
the bandwidth for the expected number of voice calls.
VoHDLC enables a Cisco MC3810 multiservice concentrator to carry live
voice traffic (for example, telephone calls and faxes) back-to-back to a
second Cisco MC3810 multiservice concentrator. VoHDLC on the
Cisco MC3810 multiservice concentrator is supported on serial ports 0
or 1, or on 0:x (the T1/E1 trunk, where x represents the
channel group number). VoHDLC traffic is carried over a serial line. As a
result, configuration is simpler than for VoIP, VoFR, or
VoATM.
