As the enterprise telephony business infrastructure continues its migration from the asynchronous transfer mode (ATM) to the Internet Protocol (IP) network technology, the transition is having an impact on packetized voice service deployment. The migration from ATM to IP introduces a fundamental challenge for designers by breaking the mandatory clock continuity between customer-premises equipment and traditional public switched telephone network (PSTN).

For optimum infrastructure usage efficiency and to deliver packetized voice services to subscribers, carriers have typically replaced traditional ATM digital subscriber line access multiplexers (DSLAMs) with IP DSLAMs, which simultaneously requires that broadband routers at each customer premise be replaced, and upgrade of private branch exchange (PBX) equipment take place. These new routers are necessary to re-establish time-division multiplexing (TDM) synchronization following the elimination of the ATM DSLAM.

There is another way to achieve this ATM to IP migration by implementing voice-over Internet Protocol (VoIP) at the source--the customer-premises broadband router--rather than in a new IP DSLAM router combination. Carriers can deploy packetized voice services while simultaneously implementing simple VoIP-based clock-synchronization alternative solutions to current, cumbersome TDM-over-IP (TDMoIP) methods. All that is required is the addition of a VoIP module in the enterprise broadband router, with no changes required to the original PBX equipment.

The benefit of the VoIP-at-the-source deployment solution is that it impacts only the customer-premises broadband router, rather than the router and the DSLAM and PBX equipment. In addition, carriers eliminate the cost of today's more expensive and cumbersome TDM emulation technology layer, while building a more sensible migration path to even better IP-based services down the road.

The lack of clock synchronization problem
Both ends of the TDM connection must be synchronized in a traditional TDM service network like the PSTN. If not, then voice frames will either be dropped (to prevent a buffer overflow condition) or inserted (to prevent an underflow condition). In both cases, connection quality and reliability suffers.

This is not a problem with traditional ATM DSLAMs, which automatically provide the means for this TDM clock synchronization (See Figure 1). In ATM-based PSTN/SDH networks, the clock master node automatically provides a time reference to secondary slave nodes. There is at least one very precise, primary reference clock within the TDM network infrastructure, and this clock has long-term accuracy defined as one part in 10¹¹, or "stratum 1." The Stratum 1 clock provides the reference clock to secondary "stratum 2" nodes, which provide a time reference to "stratum 3" nodes. The end result is that the TDM clock of the PBX at the customer premise is synchronized to the network master clock.

Figure 1. ATM-based DSLAMs provide broadband access while maintaining TDM clock synchronization

This synchronization mechanism does not exist in packet networks, nor is there any network reference available to support TDM-emulated service, or to synchronize the various TDM equipment connected to the packet network. Instead, packet networks use flow control to adapt the sending and receiving bit rate. When the ATM DSLAM is replaced with an IP-capable box, there is no longer any hardware method for the unmodified PBX and any other telecommunications equipment to access the TDM clock, which means the equipment will no longer function properly (See Figure 2). As an example, if a caller in California is speaking on the telephone to someone in France, the two clocks will not be synchronized and at some point one party will have too much--or too little--voice information and voice quality will degrade when voice frames are dropped in a random fashion to compensate for the lack of clock synchronization.

Figure 2. When the ATM DSLAM is replaced with an IP DSLAM, the timing source is removed and there is no longer a hardware link to synchronize equipment with the TDM infrastructure.

Restoring TDM clock continuity
To resolve this situation and provide a synchronization mechanism for TDM equipment inside and outside the customer premises, it is necessary to implement one of several timing schemes. A popular solution has been to use TDM-over-IP (TDMoIP), which takes TDM bit streams and converts them into IP packets for transmission over the network. The original traffic is reconstructed and the clocking is re-generated at the destination. In this scenario, the original ATM DSLAM equipment is replaced with an IP DSLAM box to deliver packetized VoIP services. Meanwhile, the customer-premise broadband router also is modified so that it can support TDMoIP for the clock-synchronization mechanism.

TDMoIP, however, requires a complete change to the broadband router platform, and inefficiently puts the entire 23-channels of the T1 TDM connection into an IP frame, even if only a single call is active. Voice compression is not used since the total T1 link is transported in its original format. Access to individual channels for re-direction, forking, snooping or other such voice applications is not possible. This reduces the flexibility of integrating in a serious fashion the communication link into the business process.

A more efficient solution is to overcome lack of TDM clock synchronization via VoIP at the source, in the broadband router on the customer premises. VoIP already has mechanisms for correcting buffer underflow/overflow conditions that are caused by TDM synchronization problems. By implementing this approach, all that is required is to add VoIP at the source (the broadband router) using a plug-in, integrated TDM-to-VoIP module (See Figure 3).

Carrier-grade VoIP firmware implements intelligent jitter buffer schemes where sophisticated algorithms are used to select the voice frames to discard at the correct moment in time to maintain voice quality. Normally, the jitter buffer is managed to increase and decrease in size during the life of the call and silence voice frames are inserted or removed at various moments to compensate for the clock offsets without the users taking notice. Voice quality is then maintained at a higher standard required by carriers.

Figure 3. In this configuration, the VoIP firmware will compensate for the clock synchronization issue.