WiMax Operator's Manual

172 CHAPTER 7 ■ SERVICE DEPLOYMENTS OVER PUBLIC WIRELESS MANS
Ensuring acceptably low latency and jitter is difficult in a pure layer 3–routed network, so
an overlay of MPLS switching or something equivalent is needed to provide the requisite QoS.
And that, as I have seen, is not always obtainable. Yet another problem arises when firewalls
must be traversed. Most firewalls filter out voice traffic because it does not provide the usual
acknowledgments necessary to keep the “pinholes” open in the firewall through which the
voice traffic must pass, so, in practical terms, a form of spoofing is required that will emulate
IP acknowledgments. Network address translation is also a problem because the ostensible
destination of the traffic is a phone number, not an IP address. Finally, it is difficult to interface
IP telephony systems with conventional billing and record keeping systems based on continuous
minutes of usage.
All these problems have been addressed by various means in current VoIP systems, but
they were not satisfactorily resolved in the prior art, hence the tardy acceptance of VoIP among
the carrier community. Today where a complete end-to-end VoIP solution is required over distance,
the carrier providing it will generally install devices called session controllers at network
exchange points and sometimes on the premises of large enterprise customers. These are used
to hand off IP traffic from one backbone to another or from an IP LAN to a WAN without any
translation to TDM circuit.
The other key component in an IP telephony system is a softswitch, which is generally separate
from the gateway. Softswitches perform the signaling functions required to transmit calls
across the PSTN, and they presuppose that the IP traffic will be translated to a TDM circuit subsequently.
In today’s telecommunications environment where as yet little end-to-end IP voice
traffic is occurring, a softswitch is probably a necessity, but I view the category as transitional
and as likely to be obsolesced toward the end of the decade.
Increasingly, IP phone systems include subscriber phone terminals that originate the
call as IP and do not require a gateway. Most such devices conform to the Session Initiation
Protocol (SIP) discussed next.
Finally, if conferencing is enabled, a separate device called a multiconferencing unit
(MCU) is generally required to connect several parties simultaneously. The central office
equipment will also need to support Universal Datagram Protocol (UDP) or, preferably, RTP.
SIP equipment provides inherent support for conferencing, so there is no longer a need for a
completely separate platform.
A rather recent development in IP telephony is the pure peer-to-peer network, which distributes
network intelligence among the subscriber nodes and requires no large infrastructure
elements such as gateways or border session controllers. Skype software is a well-known example
of the approach. At this point the benefits of this approach to the service provider are
difficult to discern because the necessity of a service provider to manage the voice calls is all
but eliminated. In the longer term, peer-to-peer IP telephony may pose a major threat to traditional
voice service models, but at present the ultimate prospects of peer to peer are a matter
of conjecture.
VoIP Standards
Telecommunications is an industry built on standards. Because ownership of network
resources is diverse, the networks have to have some means of interoperating with one
another, which necessarily entails standards. IP telephony, as it happens, has many
standards. The problem is that none is a standard in the strictest sense; that is, none is
universally accepted.