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Audio Quality Issues

If you encounter poor audio quality in VoIP communications, the source of the problem could range among many causes.

Jitter

Jitter is the variance in the intervals when Interaction Media Server receives Real-time Transport Protocol (RTP) packets. For example, if Interaction Media Server constantly receives RTP packets every 20 milliseconds, there is no jitter. If the interval in the reception of RTP packets varies, such as 20 milliseconds, 45 milliseconds, 23 milliseconds, and 50 milliseconds, this variance is jitter. Many voice over IP (VoIP) solutions use a jitter buffer to collect multiple RTP packets within a time frame so that it can queue, reassemble, and retransmit the packets with a corrected, constant interval.

Interaction Media Server uses a jitter buffer for the following VoIP communications:

  • VoIP calls that require transcoding from one codec to another.

  • VoIP calls that contain call waiting tones.

  • VoIP calls that contain intermittent tones that indicate that a call is recording.

  • VoIP calls that contain digits from a dial pad interface in CIC client software.

  • VoIP calls on CIC SIP lines with the Disable Media Server Passthru feature enabled.

For all other VoIP calls, Interaction Media Server does not use a jitter buffer and transmits the RTP packets as it receives them. For a CIC environment, this method of immediate transmission is known as a pass-through connection.

The jitter buffer for Interaction Media Server is dynamic. If Interaction Media Server detects variances in the receipt interval of RTP packets, it calculates an average receipt interval and then adjusts the jitter buffer to accumulate RTP packets within a time frame before retransmitting them. The maximum time frame for the jitter buffer in Interaction Media Server is 160 milliseconds, which equates to an average jitter of 53.3 milliseconds.

Significant variances in jitter can cause audio issues as the jitter buffer may not receive the expected number of RTP packets, which then causes Interaction Media Server to transmit silence where an RTP packet is missing. You can hear this silence as a brief break in an audio stream. Continual variances of this magnitude can produce ongoing breaks in the audio stream.

If you experience continuous breaks in audio streams, analyze each network node in the audio path and eliminate any bandwidth or processing limitations that cause jitter.

VLAN misconfiguration

If you configured Interaction Media Server to route RTP packets through the VLAN interface for data, other network nodes in the data VLAN remove any QoS DSCP markings. This problem can result in network nodes delaying the transmission of RTP packets. Also, depending on the configuration of the network, RTP packets sent to the VLAN interface for data may not be routable to the VoIP endpoint to which the system sends the packets, resulting in no audio stream for the VoIP endpoint.

Ensure that you set the RtpAddressLocal Interaction Media Server property to the correct network interface card (NIC) on the Interaction Media Server computer.

If you did not specify an address for the RtpAddressLocal property, Interaction Media Server sends the RTP packets to the first matching route in the Windows routing table.

Some NIC manufacturers include software that provides VLAN capabilities on a single NIC. If you are encountering a complete loss of audio, ensure that you configured the VLAN software for the NIC on the Interaction Media Server computer correctly. Consult the documentation for the NIC card for proper configuration.

Packet loss

When the system loses RTP packets in a network, the system also loses the portion of the audio stream within that packet. Most VoIP systems and devices play silence for that lost portion of the audio stream. The loss of RTP packets in a network can have various causes:

  • Packet degradation - If a network node cannot read the information in the IP packet containing the RTP packet, the node discards the packet. This degradation could be the result of interference along the transmission medium, insufficient quality of the transmission medium, overextension of the transmission medium, or intermittent power problems in the transmitter.

  • Long delays of single packets - Problems in a network can cause a delay of individual packets so that the system receives them long after subsequent packets. In most VoIP solutions, the system discards these delayed packets because their position in the audio stream passed and the system replaced them with either silence or sound extrapolated from the surrounding packets. Overburdened network switches, improper Quality of Service (QoS) settings in a network node, or a lack of QoS in a network node can cause delays in individual packet transmission.

Data corruption

In some instances, Interaction Media Server receives RTP packets that have corrupted data. There are several possible sources of data corruption. A probable source of data corruption is a previous network node, such as a Session Border Controller (SBC), that uses a jitter buffer and attempts to correct some perceived irregularity in the RTP packets.