[October 18, 2001]

Synchronization Essentials Of VoIP

BY PAUL SKOOG AND DOUG ARNOLD

As we accelerate into the new world of voice over IP (VoIP), we often assume we can leave some of the trappings of wireline telecom behind, such as the need for synchronization. After all, an IP network is about as asynchronous as it gets. While this may be true in some respects, precise time and synchronization continue to permeate many areas of IP telephony operations. Customer expectations of voice quality and service reliability remain unchanged, and as a result, the need for accurate timing remains unchanged though it manifests itself in different ways. The bottom line is that trying to design a VoIP network without considering network synchronization is probably the shortest path to realizing that you need it.

Voice Latency
The greatest challenge in implementing a carrier-class VoIP system is meeting the very high standard for voice quality set by the traditional Public Switched Telephone Network (PSTN) system. The same is not true in wireless phones, where inferior quality is accepted as the price of mobility.

Voice calls using VoIP technology can be made to another VoIP phone or to a traditional phone on the PSTN. Since VoIP phones currently represent only a small fraction of phones in use, we will consider the latter case in detail. The latency of the voice signal from the VoIP phone to the PSTN phone consists of delays at the following network elements:

• The VoIP telephone;
• IP network routers or switches;
• The IP to PSTN gateway;
• The wires; or
• Other delays in the PSTN system.

At the phone, the signal must be sampled, encoded, and packaged as Real Time Protocol (RTP) packets. Any routers encountered in the IP network contain input and output buffers. At the gateway, the packets will encounter more buffers (including the jitter buffer), plus delays associated with decoding and reassembling the signal. The transmission delay alone, due purely to travel time through wiring, would be about 20 ms for a call between Los Angeles and New York.

Lastly, travel through the PSTN system could involve breaking the signal up into ATM or frame relay packets and reassembling them after transport through an optical fiber. In this case, input and output buffers are encountered again. If the final destination is a VoIP phone on another network, then the phone call will have to be converted back into RTP packets, complete with encoding, decoding, and buffer delays.

Latency Measurement
Essential to assuring low latency and acceptable quality of service (QoS) and service level agreement (SLA) compliance, is the ability to make a reasonable measurement of the latency in the network. This requires adequate synchronization between measurement probes to obtain meaningful results. Some measurement solutions use hop-by-hop roundtrip metrics to build statistics on latency, jitter, etc. While this is informative, it simulates more of a piecemeal measurement approach rather than a phone conversation between two parties.

Measurement of end-to-end one way path latency better reflects a true QoS metric. An even better measurement is a one-way latency test performed simultaneously from both ends to reflect an in-process phone conversation. Synchronization on each end of the call is required for this type of test.

VoIP offers the promise of converged of voice and data, cost savings through reduction of transportation costs, and new products and services. This promise hinges in part on the customer adoption rates of VoIP, which in turn hinges on acceptable levels of quality of service. In the absence of bulletproof methods to measure, monitor, or assure quality, over-provisioning bandwidth becomes the de facto solution. Aside from increasing costs, over-provisioning only increases the probability -- without guaranteeing the quality -- of the transmission of VoIP traffic.

Network Management, Fault Diagnosis, And Recovery
Most IT organizations are measured on their ability to maintain full-flow network operations. This reliability requirement increases by orders of magnitude when you add business-related voice traffic to the same network. Any VoIP problem must be avoided, averted, or, in a worst-case scenario, minimized in an effort to keep business-critical voice systems running. To do this, one of the absolutely essential underpinnings is the accuracy of the server and router log files.

Every log file entry is time stamped. These time stamps establish the "when" of an event and as a group allow the ordering of events. Log files and subsequent reports allow you to use the log file data to identify root cause problems within your network. Since server logs are a compilation of information from different hosts, it is essential that the time stamps be correct. If not, you will have difficulty ordering events and troubleshooting root-cause problems. The more difficulty you have in identifying a problem, the longer the QoS level of your VoIP system may degrade,  or worse yet, be non-operational.

Figure 1. Time-accurate server and router log files play a key
role in troubleshooting and identifying root cause problems.

Time synchronization across network servers, routers, and network devices is not a difficult endeavor. Using the well-established Network Time Protocol (NTP, RFC 1305) and a reliable time source, such as a dedicated network time server that references the GPS system, synchronization of servers and network devices can be easily maintained. In fact, many operating systems and network devices already incorporate support for NTP.

NTP uses Coordinated Universal Time (UTC), which is the same worldwide. The GPS satellite system is the most readily-available source for UTC time in the world. By synchronizing your network to UTC, you remove one more source of interoperability problems between your network and others. This is important since VoIP traffic may transit many networks, requiring the correlation of log files from various networks to solve a problem. Time servers on the market today provide accurate and secure time and are able to synchronize thousands of clients on the network.

Call Detail Records Need To Be In Sync
No discussion regarding time on a VoIP network would be complete without mention of the obvious role time plays in billing. Call Detail Records (CDR) provide information about call origination, destination, and duration. Duration certainly includes the time stamp when the call was initiated, and either the call duration or time the call was terminated. Billing integrity will rely on the underlying time accuracy of the VoIP CDR records. Without proper synchronization, the CDR accuracy will falter and the billing system will inevitably come into question. This is particularly critical when the CDR information is shared between carriers and billing discrepancies require time-consuming mediation.

Synchronizing The Gateway Interface
Eventually, packets arrive at the gateway between the VoIP system and the PSTN. The PSTN uses a very defined timing hierarchy for synchronization of traffic on the network. PSTN voice packets must arrive in order, and with low latency and jitter. This gateway also represents a change in the general synchronization requirements: VoIP systems synchronize by way of time stamps to aid in latency reduction and network log file integrity; the PSTN uses synchronization to improve efficiency and data throughput.

Figure 2. Precise time is required for VoIP;
precise frequency is required for PSTN.

Providing this synchronization requires a versatile time reference that can supply the Stratum 1 level frequency reference for the PSTN and the accurate time stamps for the VoIP side. Many of these VoIP/PSTN gateways, also known as a softswitches, already employ NTP for accurate time stamping. Stratum 1 level timing is already permeating the edge the PSTN network by synchronizing customer premise ATM routers and switches. By adding VoIP to the edge, it further increases the need to expand the synchronization capabilities of the Stratum 1 timing clock. Again, a quality GPS-referenced clock can support both the NTP and the Stratum 1 time and frequency requirements.

Sync Now Or Sink Later
Synchronization is usually far from top priority when establishing a VoIP network. However, as soon as quality problems occur, the true value of the synchronization system becomes very clear. QoS monitoring systems and network diagnostic programs will ultimately drive the requirement for synchronization across network routers, servers, and related devices. These systems rely on log file accuracy and integrity for their metrics. Without time stamp accuracy, an unacceptable amount of time will be spent trying to resolve problems that could have been avoided, or cleared up more efficiently. Good practices consider synchronization important enough to incorporate up-front in system design, rather than later when trouble occurs.

Paul Skoog is the product manager for IP Networking Products, and Doug Arnold is the chief scientist for TrueTime, Inc. TrueTime designs and manufactures precision time and frequency products for network and telecommunications synchronization.