This section reviews the requirements of the PSTN with regards to a VoIP application as well as the timing issues that are critical for toll quality voice deployment.
How VoIP over WLAN applications will be deployed will have an effect on the design and integration of the equipment. The following issues have a bearing on equipment design:
- VoIP voice compression algorithm(s)
- Voice packet size, packet rate and delay
- Timing requirements for signaling and call set up
- Call control protocol
- Capacity and range of QoS capabilities that will be supported beyond voice
The following figure illustrates enterprise deployments:
VoIP integration in the enterprise will be evolutionary, not revolutionary. Greater than 90 percent of enterprises use analog and ISDN (P-Phone)-based PBX equipment. In some cases the PBX equipment is modular and supports VoIP interface LRUs (line replaceable units). For older legacy PBX equipment, a VoIP gateway can introduce VoIP into the enterprise.
Given the expense of analog PBXs and phones, enterprises typically decide to gradually transition the deployment of VoIP. The enterprise deployment diagram above illustrates a phased deployment such as the following:
- The analog PBX/voice mail system connects to the PSTN. A small number of users have VoIP phones connected via a VoIP gateway. (See Hybrid Network-1.)
- A VoIP PBX/gateway connects the enterprise to the PSTN. A majority of users have VoIP phones. Legacy phones are supported by an IP/PBX converter. This network will be typical in larger enterprises with a large number of PSTN connections. (See Hybrid Network-2.)
- The enterprise has an IP connection to a remote gateway/PBX that may serve one or more business customers. Legacy analog phones, if any, are supported by an IP/PBX converter. This network will be typical in small and medium size enterprises and in the remote facilities of large enterprises. (See IP Network.)
A residential network topology, as shown in the following diagram, will have its own set of challenges.
Residential/SOHO customers typically have one or more analog voice circuits which are connected directly to the PSTN through a CLASS 5 switch or through a DLC remote terminal. In many cases, subscribers are served by advanced DLC/DSLAM (DSL access multiplexer) remote terminals that provide a combination of DSL and POTS service on a single line as well as standard POTS voice circuits.
As the PSTN network has been upgraded to support DSL, integrated voice/data service (VoDSL) using ATM AAL1/2 packet voice has become available. In general, most residential and SOHO users will continue to have POTS as the primary voice interface. The following are several of the interface configurations that will emerge for home users:
- POTS voice interfaces to the PSTN in a way that is identical to a cordless phone. VoIP conversion to/from the analog wire pair would be integrated into the VoIP-enabled access point.
- VoDSL interfaces to the PSTN and VoATM is converted to VoIP for use with a WLAN AP.
If the residential/SOHO user is interfacing to a broadband cable modem, the topology is nearly identical to that of DSL. Where voice over cable is available, it is typically an independent system from the DOCSIS® cable modem interface. An example of this independent interface is the ARRIS system, which results in termination of two-wire analog POTS interfaces at the customer premise. As DOCSIS 1.1 and 2.0 become readily available, integrated VoIP over cable modem will become prevalent.
The time delay of the communications path for 802.11 VoIP in a cordless residential application will typically be much shorter and have a limited number of sources of delay. The voice interface to the home will typically be a POTS interface. Specific signaling requirements must be supported with analog line pairs to DLC and/or CLASS 5 switching equipment. (See note below.)
The design of DLC and Class 5 switch equipment is based on the premise that the wire connection to the phone is in place and operational. Typically, establishing a call with Q.391 protocol is completed in less than 200 msec. The caller ID modulation receiver (FSK Modem) must be ready after the first ring, which is less than or equal to two seconds. Worst case signaling delay should be less than 100 msec.
This creates a problem.
Any WLAN implementation will include VoIP handsets that are often in power-saving sleep modes where much of the device is not operating. In this mode, the WLAN will "wake up" and establish communication in intervals of 200 msec to one sec, well beyond the telephony system delay specification.
Clearly, the 802.11 AP will have to maintain an attached state for a given handset and provide a call proxy during call set up until the call can be handed off to the handset after it has awakened from sleep mode.
