Calls using public VoIP services have become quite common. When the call is free as with Skype, or very cheap with Vonage, the callers may accept slightly less call quality than the PSTN calls. But call center callers and agents need good-quality voice calls to transact business.
To learn about this issue, I spoke with Matthew Able, Lead Sales Engineer and Joe Hamilton, Manager, Enterprise Operations, both at IntelePeer, a SIP trunking provider. Matthew has observed Internet network performance for years and has seen it improve, producing lower packet loss rates, less jitter and shorter latency (end-to-end delay), thereby benefiting voice communications services offered through Internet access lines.
Voice quality is very important for call center agents and customers. Poor voice quality leads to the perception that the call center represents low value to the organization. Poor quality can lead to disappointed or frustrated customers, transaction errors and lengthened call times, all of which reduce profitability.
The SIP trunk, carrying digitized voice, has now become the preferred (if not yet widely deployed) technique for accessing the PSTN. There are two common digital voice standards: G.711, 64-kbps uncompressed voice, and G.729 8-kbps compressed voice. G.711 is most commonly implemented for call center voice communications since it delivers higher voice quality, requires less complexity to implement and can reduce transcoding (conversion) delay. The total bandwidth required including packet overhead for G.711 is about 80 kpbs. The G.729 bandwidth requirement is about 24 kbps including the packet overhead. The actual bandwidth depends on the SIP trunking provider's implementation and can be greater or less.
The common standard for measuring voice Quality is Mean Opinion Score (MOS). The MOS ranges from 1 (bad) to 5 (excellent). According to Joe Hamilton, G.711 delivers an average MOS of 4.1.; G.729 delivers a MOS slightly better than 3.9. Both MOS values deliver acceptable quality, but most call centers prefer the MOS to average 4.1 or better.
The SIP trunk subscriber should independently monitor the voice quality as well as read the provider's MOS reports. Sometimes the providers measure the voice quality over long periods of time that can produce an average voice quality that is not reflective of the voice quality during the call busy hour.
One of the recommendations of IP Telephony vendors is to implement Quality of Service (QoS) over the enterprise network to ensure voice quality. But once the VoIP outbound transmission passes to and through the Internet, QoS is not supported. The enterprise can implement QoS for voice through the Session Border Controller (SBC) connected to the SIP trunk. But once the voice packet is on the Internet, voice packets are treated equally with the same as data packets. Inbound transmission will have lost any QoS capability when packets are received from the Internet.
Matthew and Joe recommended that the call center ensure that there is enough bandwidth to handle the calls over the SIP trunk. It is recommended that the SIP trunk operate below an average of 80% utilization. It is also recommended that the voice and data access over the SIP trunk be logically or physically separated to eliminate bandwidth contention between voice and data packets.
According to Matthew and Joe, IntelePeer has been able to deliver consistent voice quality. Organizations that have implemented SIP trunking for voice calls stayed with the SIP implementation, and IntelePeer has not encountered subscriber turnover.
A few years ago this might not have been achievable because of the poorer Internet performance. Today, the Internet is delivering the necessary performance. SIP trunking to the PSTN will continue to grow because of the voice quality track record as well as the cost reduction when compared to T1 and PRI PSTN access.
