Considerations When Integrating Cisco CME in H.323 and SIP VoIP Networks

H.323 is the dominant protocol deployed for VoIP networks from an installed-base perspective. Because H.323 is more mature than SIP, you can expect to see increased real-world interoperability between different vendors' H.323 products, particularly with basic call handling. However, many of the high-level VoIP networking considerations that apply to H.323 apply equally in the SIP context. Some technical and protocol-specific differences exist between H.323 and SIP VoIP networking, but for the most part, you'll find more commonality than difference, at least at the level of technical detail that this chapter addresses.

The shared aspects of the two protocols means that the overall high-level architecture and distribution of hardware and primary component roles within your VoIP network don't significantly depend on which protocol you choose to use for intersite VoIP. For networks built on either H.323 or SIP, you are dealing with peer-to-peer communication between sites. Therefore, you also need some kind of telephone number directory system to be able to resolve the IP address of the appropriate destination VoIP peer device for intersite calls.

In contrast, this similarity between H.323 and SIP does not extend to Media Gateway Control Protocol (MGCP) (and also Skinny Client Control Protocol [SCCP]), which takes a significantly different approach to telephony. Of course, it is still possible to connect Cisco CME to MGCP networks, primarily using either H.323 or SIP. Many MGCP Call Agent implementations (using MGCP internally for phone control) use H.323 or SIP to connect separate Call Agents (as intersystem peer-to-peer). Cisco CME itself does not support control of MGCP endpoints. Cisco CME uses SCCP for phone control, and SCCP shares many common traits with MGCP.

The term VoIP here specifically describes "long-distance" VoIP telephone calls that traverse a WAN. This interpretation excludes SCCP used to control local IP phones. Although SCCP technically does use VoIP technology, it is primarily used in the context of operating voice calls within the confines of a LAN with more or less unlimited bandwidth and many fewer concerns about security.

You can view the H.323/SIP versus SCCP contrast as the difference between interbranch office voice traffic and intrabranch office voice traffic, or alternatively as long distance (WAN) versus local VoIP (LAN). This division is useful in many ways, because it inherently supports the often-necessary difference in treatment of calls between internal and external phone users.

In some cases, you will want to treat H.323 calls as internal calls and won't want a high degree of differentiation in the treatment of LAN versus WAN calls, such as calls between separate systems on two floors of the same building. Cisco CME has features that address this, although currently you cannot treat a network of many Cisco CME systems as if they are a single logical entity with full intersite feature transparency. Both H.323 and SIP still have obstacles to overcome before this is really possible. Not least of these are issues surrounding meaningful interoperability with non-Cisco devices for services beyond basic calls.

When you extend VoIP calling into the WAN space, you might also have to consider the difference between VoIP calls that come from other Cisco CME nodes within your WAN network versus VoIP calls that are from VoIP Public Switched Telephone Network (PSTN) gateways or even from other independent external/wholesale VoIP carrier networks. You can link independent VoIP networks together and into your corporate VoIP network using IP-to-IP gateways. This arrangement may be desirable if you want to obtain international and long-distance phone service directly from a carrier-class VoIP service provider and have this linked at the VoIP level to your private enterprise VoIP network.

SIP potentially has some advantages over H.323 in terms of separating intersite VoIP calls from true external VoIP calls, because SIP uses the Internet concept of domains. It's a fair assumption that all of the intersite calls will use the same root domain name and that this fact can be used to make the required distinction. However, from a purely practical security point of view, most likely you will want any truly external VoIP traffic entering your corporate VoIP network to pass through an IP-to-IP gateway and also a firewall, regardless of whether you choose to use SIP or H.323. This means that you should have the opportunity to appropriately classify and mark the external calls at the point of entry in either type of network.

Alternatively, you can keep your VoIP network entirely separate at the IP level and simply connect into VoIP service provider carrier networks through time-division multiplexing (TDM)-based PSTN-like gateways (at some cost in terms of increased end-to-end voice path delay). For the sake of simplicity and clarity, the rest of this chapter ignores the IP-to-IP possibility and includes only the PSTN gateway scenario. For many reasons, what is on the far side of the gatewaywhether PSTN or IP-to-IPisn't hugely significant. It's the gateway's job to take care of whatever adaptation is needed to provide the interconnection path.

Integrating Cisco CME in an H 323 Network

Cisco IP Communications Express(c) CallManager Express with Cisco Unity Express
Cisco IP Communications Express: CallManager Express with Cisco Unity Express
ISBN: 158705180X
EAN: 2147483647
Year: 2006
Pages: 236
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