Support for protocols, signaling capabilities, voice features, and voice applications is changing and growing quickly. You must have a good understanding of voice network architectures to know which business requirements each architecture addresses. Also, gateways play an important role in providing access to the right mix of functionality. You must understand the main features and functions required in enterprise and service provider environments to choose the appropriate gateway.
Centralized Network Architectures
One benefit of VoIP technology is that it works with centralized and distributed architectures. This flexibility allows companies to build networks characterized by both simplified management and endpoint innovation. It is important to understand the protocols that are used to achieve this type of VoIP network agility.
The multisite WAN model with centralized call processing, as illustrated in Figure 5-2, consists of the following components:
- Central gateway controller (call agent) The call agent handles switching logic and call control for all sites under the central controller. A central gateway controller includes both centralized configuration and maintenance of call-control functionality. When new functionality needs to be added, only the controller needs to be updated.
- Media gateways Media gateways provide physical interconnection between the telephone network, individual endpoints, and the IP network. Media gateways communicate with the call agent to notify it of an event. An example is a telephone going off hook. The gateway also expects direction from the call agent on what action to take as a result of the event. For example, the call agent tells the gateway to provide a dial tone to the port that sees the off-hook condition. After the call-control exchange is completed, the gateways route and transmit the audio or media portion of the calls. This is the actual voice information.
- IP WAN The IP WAN carries both call control signaling and voice payload between the central site and the remote sites. QoS configuration is highly recommended when voice packets are transported across a WAN to ensure that the voice packets get priority over data packets in the same network. To minimize bandwidth use for voice streams that are crossing the WAN, the G.729 coder-decoder (CODEC) is used to compress the voice payload. G.729 compresses voice to 8 kbps per call as opposed to the 64 kbps traditionally used in LAN and PSTN environments.
Centralized call-processing deployments typically offer the following characteristics:
- MGCP or Megaco/H.248 protocol for call control
- Cisco CallManager at central site for managing call control
- Centralized applications pointed to by remote sites
- Up to 30,000 IP phones per cluster
- Call Admission Control (CAC) to limit number of calls per site
- Survivable Remote Site Telephony (SRST) for remote branches
A typical use for centralized architecture is a main site with many smaller remote sites. The remote sites are connected via a QoS-enabled WAN but do not require full features and functionality during a WAN outage. MGCP and Megaco/H.248 are the prevalent signaling protocols used in centralized architectures to control gateways and endpoints.
Applications such as voice mail and IVR systems are typically centralized to reduce the overall cost of administration and maintenance.
Cisco CallManager clusters can support up to 30,000 IP phones per cluster, providing for a scalable solution in enterprise environments. For even more scalability, clusters can be interconnected via intercluster trunks.
CAC is administered by the Cisco CallManager cluster. CAC is critical in enterprise implementations that include WAN connections because these connections typically have limited bandwidth that is shared between voice and data users. Control must be established over the number of calls that can flow concurrently across the WAN at any given time so that as the call volume grows, overall call quality does not diminish.
One disadvantage of implementing a centralized architecture is that if the WAN connection fails between the remote site and the central site that houses Cisco CallManager, no further voice calls can be processed by the remote site. Additional steps need to be taken to ensure that data and voice services at the remote sites remain available. One option is to implement redundant WAN links between the remote sites and the central site. In many cases, this solution is not financially feasible. Alternatively, Survivable Remote Site Telephony (SRST) provides high availability for voice services. SRST provides a subset of the call-processing capabilities within the remote-office gateway. It also enhances the IP phones with the ability to "re-home" to the call-processing functions in the local gateway if a WAN failure is detected. This feature allows the remote site to continue to provide voice connectivity in the absence of the WAN link.
Most centralized VoIP architectures use MGCP or Megaco/H.248 protocols. You can also build session initiation protocol (SIP) or H.323 networks in a centralized fashion. This is done using back-to-back user agents (B2BUAs) or gatekeeper-routed call signaling (GKRCS), respectively.
Figure 5-2 shows a typical centralized call-processing deployment, with a Cisco CallManager cluster acting as the call agent at the central site and an IP WAN with QoS enabled to connect all the sites. The remote sites rely on the centralized Cisco CallManager cluster to handle their call processing but have local voice-enabled routers to perform the voice-gateway translations for media streams. Each remote site connects locally to the PSTN. Long-distance (LD) service might be provided from the head office or through each local PSTN connection.
H.323 Distributed Network Architectures
The multisite WAN architecture with distributed call processing consists of multiple independent sites. Each site has its own call-processing agent, which is connected to an IP WAN that carries voice traffic between the distributed sites.
Each site in the distributed call-processing architecture using H.323 can be comprised of one of the following:
- A single site with its own call-processing agent
- A centralized call-processing site and all its associated remote sites
- A legacy PBX with a VoIP gateway
Figure 5-3 provides an example of a distributed call-processing architecture. Notice, in the figure, that each site contains its own call-processing agents (in the form of CallManager clusters). This type of call-processing approach has the following characteristics:
- No call control signaling for intrasite and off-net calls through the IP WAN
- Transparent use of the PSTN if the IP WAN is unavailable
- Logical hub-and-spoke topology for the directory gatekeeper
- Only one type of CODEC configured for the IP WAN
Multisite distributed call processing allows each site to be completely self-contained. The IP WAN in this model does not carry call-control signaling for intranet and off-net calls because each site has its own Cisco CallManager cluster. Typically, the PSTN serves as a backup connection between the sites in case the IP WAN connection fails or does not have any more bandwidth available.
Distributed architectures are associated with H.323 and SIP protocols. These protocols allow network intelligence to be distributed between endpoints and call control devices. Intelligence in this instance refers to any aspect of call handling including the following:
- Call state
- Calling features
- Call routing
- Provisioning
- Billing
The endpoints can be VoIP gateways, IP phones, media servers, or any device that can initiate and terminate an H.323 VoIP call. The call control devices are called gatekeepers (GKs) in an H.323 network. In an enterprise environment where many gatekeepers are required, a second level of hierarchy is achieved through the use of directory gatekeepers (DGKs). Directory gatekeepers provide summarization capabilities for multiple configured gatekeepers.
The multisite WAN architecture with distributed call processing consists of the following components:
- Media gateways Media gateways provide physical interconnection between the telephone network, individual endpoints, and the IP network. The media gateway translates call signaling between the PSTN or local endpoints and the IP network. The media gateway must contain the call-processing intelligence to perform all call-handling functions related to H.323. Media gateways communicate with gatekeepers for call address resolution and CAC.
- Gatekeeper A gatekeeper is an H.323 device that provides CAC and E.164 number resolution. Gatekeepers are among the key elements in the multisite WAN model that have distributed call processing. Gatekeepers provide dial plan resolution, which improves scalability in an H.323 network. Without gatekeepers, each gateway would need to be configured to know where all other reachable telephone numbers were located. The gatekeeper provides a central repository of telephone numbers and the gateways associated with those numbers. When the network is configured for gatekeepers, the learning process is dynamic, because all participating gateways register with the gatekeeper and notify it of available telephone numbers. The gatekeeper also provides CAC to ensure that voice quality is not diminished when a large number of calls enter the network.
- IP WAN The IP WAN carries call control signaling and voice payload for intersite voice communication only. Call signaling and voice transmission for all intrasite calls and off-net calls that are going to the local PSTN remain local to the site. QoS configuration is highly recommended when voice packets are transported across a WAN, to ensure the voice packets get priority over data packets in the same network. As in the centralized system, to minimize bandwidth use for voice streams that are crossing the WAN, the G.729 CODEC is typically used to compress the voice payload. G.729 compresses voice to 8 kbps per call as opposed to the 64 kbps that is traditionally used in LAN and PSTN environments.
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