VoIP or any real-time application are delay sensitive. One second network delay can impact several seconds of conversation. Broadly, VoIP degradation is the result of three major culprits:
1> Network Latency
2> Jitter
3> Packet Loss
Network Latency: It's the delay for the packets to travel from the speaker to the listener. If network latency is above 150ms on your VoIP environment, it will result in huge degradation in voice quality. More than 90% of time, your ISP is the culprit for high network latency.
Jitter: Delta in the end-to-end arrival time between the received packets. It's resulted because of varying network traffic and conditions on VoIP environment. Jitter below 25 ms is acceptable. Jitter Buffer can be adjusted to reduce the impact of jitter on voice quality. However Jitter Buffer can only handle jitter level up to 100ms.
Packet Loss: It is resulted of heavy network load and congestion. In VoIP environment, retransmission doesn't make any sense as conversation is real-time. Merely 10% (non-consecutive packets) of Packet Loss can cause serious impact on voice quality. Since VoIP uses UDP, it doesn't allow to monitor packet loss detection [ not quite true; RTCP should be able to tell packet loss]
Voice Quality is measured using two widely used factors: MOS (Mean Opinion Score) and R-value. MOS is measured by group of listeners scaling the quality from 1(unintelligible) to 5 (very clear) (defined in ITU P.800). R-value uses mathematical formula incorporating Network Latency, Jitter and Packet Loss and grades on the scale of 1(unintelligible) to 100(very clear) (defined in ITU-T G.107)
R-value-------MOS-------Remarks
90-100-------4.2+--------Very Satisfied
80-90------4.0-4.3-------Satisfied
70-80-------3.6-4.0------Some Unsatisfied
Wireshark can be used to pull the Network Latency and Jitter information from the packet capture done on your VoIP environment.
Access Voice quality SLA from Verizon
References:
Hacking VoIP Exposed by David Endler & Mark Collier
www.verizonbusiness.com
1> Network Latency
2> Jitter
3> Packet Loss
Network Latency: It's the delay for the packets to travel from the speaker to the listener. If network latency is above 150ms on your VoIP environment, it will result in huge degradation in voice quality. More than 90% of time, your ISP is the culprit for high network latency.
Jitter: Delta in the end-to-end arrival time between the received packets. It's resulted because of varying network traffic and conditions on VoIP environment. Jitter below 25 ms is acceptable. Jitter Buffer can be adjusted to reduce the impact of jitter on voice quality. However Jitter Buffer can only handle jitter level up to 100ms.
Packet Loss: It is resulted of heavy network load and congestion. In VoIP environment, retransmission doesn't make any sense as conversation is real-time. Merely 10% (non-consecutive packets) of Packet Loss can cause serious impact on voice quality. Since VoIP uses UDP, it doesn't allow to monitor packet loss detection [ not quite true; RTCP should be able to tell packet loss]
Voice Quality is measured using two widely used factors: MOS (Mean Opinion Score) and R-value. MOS is measured by group of listeners scaling the quality from 1(unintelligible) to 5 (very clear) (defined in ITU P.800). R-value uses mathematical formula incorporating Network Latency, Jitter and Packet Loss and grades on the scale of 1(unintelligible) to 100(very clear) (defined in ITU-T G.107)
R-value-------MOS-------Remarks
90-100-------4.2+--------Very Satisfied
80-90------4.0-4.3-------Satisfied
70-80-------3.6-4.0------Some Unsatisfied
Wireshark can be used to pull the Network Latency and Jitter information from the packet capture done on your VoIP environment.
Access Voice quality SLA from Verizon
References:
Hacking VoIP Exposed by David Endler & Mark Collier
www.verizonbusiness.com
