A lab test shows the differences between TDM and VoIP conferencing bridges.

Channel Partners

September 28, 2010

4 Min Read
Testing Conference Call Quality

By Herb Levitin

Cell phone use has lowered users expectations for voice quality for calls except for conference calls. Clients will complain about voice quality during a conference call, especially when the conference call is recorded. Their complaints can be linked to the migration from older TDM (Time Division Multiplexing) to newer VoIP technology. Technically you can get better than toll-grade speech quality using high-bandwidth VoIP technology, but you also can get worse quality.

To prove the point, I hired an outside consulting firm, GL Communications Inc., to test eight different conferencing bridges to compare quality. The results are summarized in the table, but generally showed that voice quality and delay varied widely among the bridges tested. The connection type also had an impact. The best sounding bridge used the oldest technology and the older TDM telephone connection. (The full report can be downloaded from www.wholesaleconference.com/GLVQTDocument.pdf)

The Bridges. The workhorse TDM bridges use digital signal processors (DSPs). They have adapted to receiving VoIP traffic by keeping the DSPs and replacing the TDM T1-PRI ports with an Ethernet card. The next-generation teleconference bridges were designed from scratch to be VoIP bridges. For backwards compatibility some bridges have a DS3-TDM port, but they are doing an immediate conversion from TDM to VoIP and all processing is done with VoIP packets, which introduces an additional delay versus the older DSP-based TDM bridges.

The Connections. The majority of PSTN traffic carried by interexchange carriers has migrated to VoIP but has retained the same 64kbps channels as the older TDM circuits to minimize speech quality issues. The same 64kbps channel of TDM takes more than 80kbps of bandwidth with the extra overhead required by VoIP. The real advantage of VoIP versus TDM for the IXC is the majority of the time there is no information being sent due to the nature of human speech. During the time a conversation is silent many other conversations are carried over the same circuit. In contrast, TDM is limited to one conversation.

The majority of PSTN traffic carried by interexchange carriers has migrated to VoIP but has retained the same 64kbps channels as the older TDM circuits to minimize speech quality issues. The same 64kbps channel of TDM takes more than 80kbps of bandwidth with the extra overhead required by VoIP. The real advantage of VoIP versus TDM for the IXC is the majority of the time there is no information being sent due to the nature of human speech. During the time a conversation is silent many other conversations are carried over the same circuit. In contrast, TDM is limited to one conversation.

The IXC networks are designed to have a maximum quality of service (QoS) with minimum delays and congestion. In many facilities the incoming PSTN is on TDM circuits that are connected to a softswitch that converts the incoming TDM to VoIP but only travels a few feet to a local VoIP bridge. There is minimal delay or speech quality degradation introduced at the local facility during this conversion. There also are no variable delays caused by running VoIP traffic over the Internet as long as the traffic remains inside the facility.

If any portion of the route the VoIP traffic takes is using the Internet, then all bets are off. Without control over packet delay, its difficult to control speech quality. Another way speech quality is degraded is changing the VoIP protocol to use a lower sampling rate. Going from a 64kbps protocol such as G.711 to the 8kbps protocol such as G.729 changes the Ethernet bandwidth per call from 87.2kbps to 31.2kbps. You can hear the difference between these two protocols. In addition, VoIP bridges processing levels can cause variable delays that impact speech quality. In contrast TDM bridges using DSPs process less and on a synchronized basis so delays are always fixed.

Aside from using an independent test lab, the best way to determine if a provider is offering high-quality teleconferencing is to first ask questions about what type of bridge and network connections it is using. Then make multiple test conference calls using analog telephone lines; speak into one telephone while listening to the other line. Speech quality is subjective so test as many services as possible.

Herb Levitin is the founder of Wholesale Conference, a Santa Barbara, Calif., provider of audio and Web conferencing services. He can be reached at [email protected].

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