IP telephony solution. Gatekeepers can be used for
scalability of dial plans and for bandwidth management when
using the H.323 protocol. Figure shows a company with a
headquarters in Chicago, two offices in the San Francisco area,
and one smaller office in Dallas. In Chicago, three
headquarters locations connect via a metropolitan area network
(MAN). The main west coast office in San Francisco connects
with the San Jose office via a MAN. Dallas has a single site.
The three main locations of the company (Chicago, San Francisco
Bay Area, and Dallas) interconnect via an IP WAN. The Chicago,
San Francisco, and Dallas locations each have a Cisco Unified
CallManager cluster serving local IP phones and IP phones
located in the MAN-connected sites. At the airport campus in
Chicago, IP phones are not used because the company is using a
managed telephony service that is offered by the owner of the
building. However, a voice gateway connects to the managed PBX,
allowing VoIP calls to and from the airport office telephones
through the gateway. The Chicago headquarters sites use the IP
WAN router as a gatekeeper that provides CAC and bandwidth
management for H.323 calls. In addition, each main site has a
voice gateway that connects to the PSTN, allowing off-net
calls. These gateway routers are equipped with digital signal
processors (DSPs) that provide conferencing and transcoding
resources. Within each area, the G.711 codec is used, while
calls between the three areas use the G.729 codec. All calls
within the enterprise should use the IP WAN. If the IP WAN
fails, or if calls are denied by CAC, the calls are rerouted
through the PSTN.
Content 2.5 Implementing VoIP
in an Enterprise Network 2.5.2 Deploying CAC
IP telephony solutions offer CAC to limit the number of
concurrent voice calls permitted in order to prevent
oversubscription of WAN resources. Figure shows how CAC is used
in a network. Without CAC, if too many calls are active and too
much voice traffic is sent at once, delays and packet drops
occur. Even giving RTP packets absolute priority over all other
traffic does not eliminate problems when the physical bandwidth
is not sufficient to carry all voice packets. QoS mechanisms do
not associate individual RTP packets with individual calls;
therefore, all RTP packets are treated equally. All RTP packets
will experience delays, and any RTP packets can be dropped. The
effect of this behavior is that all voice calls experience
voice quality degradation when oversubscription occurs. A
common misconception is that only calls that are beyond the
bandwidth limit suffer from quality degradation. CAC is the
only method that prevents general voice quality degradation
that is caused by too many concurrent active calls. Example:
CAC Deployment
Figure shows a scenario with two sites,
with three telephones at each site connected to a VoIP gateway
through a PBX. The two gateways are connected via an IP
network. The network is designed for a maximum of two
concurrent calls. If CAC is not used, whenever there are three
active calls, all three calls experience severe voice quality
degradation. When CAC is deployed, the gateways are configured
to allow no more than two calls at the same time. When a third
call is attempted, the call is blocked. With the CAC
configuration, no voice quality problems should be
experienced.
Content 2.5 Implementing VoIP in
an Enterprise Network 2.5.3 Voice Gateway
Functions on a Cisco Router Cisco routers, especially the
Cisco Integrated Services Router (ISR), such as the Cisco 3800
Series ISR, are voice capable. These routers can be equipped
with traditional telephony interfaces to act as gateways for
analog and digital devices including telephones, faxes, PBXs,
and the PSTN, allowing those devices to interact with VoIP
networks. Figure lists voice gateway functions in Cisco
routers. The routers support numerous analog interfaces,
digital interfaces, and signaling protocols: - Foreign
Exchange Station (FXS)
- Foreign Exchange Office
(FXO)
- Ear and Mouth (E&M)
- T1 or E1
channel associated signaling (CAS) and T1 or E1 common channel
signaling (CCS) using ISDN
- Q Signaling (Q.SIG)
protocols
Gateways with analog interfaces convert
analog signals into digital format before encapsulating voice
into IP packets. The gateways can compress digitized voice
before the encapsulation happens. This compression reduces the
bandwidth that each call needs. Cisco IOS routers support
H.323, session initiation protocol (SIP), and Media Gateway
Control Protocol (MGCP) for VoIP signaling. In addition,
gateways can be equipped with DSPs, which provide conferencing
and transcoding resources. In IP telephony environments,
gateways support fallback scenarios for IP phones that have
lost IP connectivity to their call agent (that is, Cisco
Unified CallManager). This feature, called Cisco Survivable
Remote Site Telephony (SRST), enables the gateway to take the
role of the call agent during WAN failure. Local calls can then
proceed even if IP connectivity to Cisco Unified CallManager is
broken. In addition, Cisco SRST can route calls out to the PSTN
and, thus, use the PSTN as the backup route for calls toward
any site that is not reachable via IP. Further, Cisco IOS
routers can permanently act as a call agent for IP phones. The
feature that provides this functionality is Cisco Unified
CallManager Express. With Cisco Unified CallManager Express,
the router provides Cisco Unified CallManager functionality. If
the router is also a voice gateway, the router combines IP
telephony and VoIP gateway functionality in a single box. Cisco
IOS gateways also support other features, such as call
preservation (Real-Time Transport Protocol [RTP] stream) in
case of a lost signaling channel, dual tone multifrequency
(DTMF) relay capabilities, supplementary services support (for
user functions, such as hold, transfer, and conferencing), and
fax and modem support.
Content 2.5
Implementing VoIP in an Enterprise Network 2.5.4
Cisco Unified CallManager Functions Cisco Unified
CallManager is the IP-based PBX in an IP telephony solution.
Cisco Unified CallManager acts as a call agent for IP phones
and MGCP gateways and can also interact with H.323 or SIP
devices using the devices’ protocols. For redundancy and load
sharing, multiple Cisco Unified CallManager servers operate in
a cluster as shown in Figure . From an administration
perspective, the whole cluster is a single logical instance.
There are six main functions performed by Cisco Unified
CallManager: - Call processing: Cisco Unified
CallManager processes calls between end devices and gateways.
Call processing includes call routing decisions, signaling
between the affected devices, and accounting of calls. In
addition, class of service (CoS) and bandwidth management can
be configured to influence call processing.
- Dial
plan administration: Cisco Unified CallManager acts as a
call agent for IP phones and MGCP gateways and thus eliminates
the need for local call routing tables in these devices. Only
the call agent (that is, Cisco Unified CallManager) needs to
know the dial plan. This means all dial plan administration is
performed on the Cisco Unified CallManager. H.323 and SIP
devices follow the distributed call-processing model and
require locally available and administrated dial plans.
- Signaling and device control: In the role as a call
agent, Cisco Unified CallManager controls IP phones and MGCP
gateways by telling these devices what to do in certain events.
For instance, when an IP phone informs Cisco Unified
CallManager that the user has gone off-hook, Cisco Unified
CallManager tells the IP phone to update the screen and play a
dial tone.
- Telephone feature administration:
The entire IP phone configuration is entered and stored at
Cisco Unified CallManager. The IP phones load their
configuration file during boot after a device reset. IP phone
administration is fully centralized.
- Directory and