Spirent's Audio Quality Expert discusses the advantages of implementing an EVS Audio Codec on wireless devices.
Rollouts of high-bandwidth 4G LTE networks are accelerating globally, providing consumers with access to evolving data-based applications such as augmented reality and multi-player gaming. In addition to these advanced data services, consumers expect access to traditional voice services. Voice and video over LTE (VoLTE/ViLTE) is the tool to provide this integration, with a roadmap of upcoming features including advanced services such as video calls, presence advertisement, and RCS.
As VoLTE is deployed into LTE networks, consumers expect the performance of these services to be at least equal to traditional circuit-switched services. Since LTE carries voice as an IP packet service, these data streams are subject to numerous network impairments, including packet loss, delay, and jitter. In addition, registration, authentication, signaling compression, security, and call setup times have the potential to take longer than circuit-switched networks due to the complexity added by the IMS Core network servers, collectively known as the Call Session Control Function. Device performance must be tested and optimized in the lab with a high quality simulated LTE/IMS network to guarantee customer satisfaction with VoLTE services.
It seems that everyone knows by now that proper network conditions are necessary for a high quality experience in audio communications. And so I’ve been asked numerous times regarding the requirements from the network. What are the minimum network conditions to allow for a high audio quality experience? Can you suggest thresholds for packet loss and delay?
Packet loss is considered to be the most annoying disturbance in interactive voice communications. In fact, in a poll question posted in a webinar on user experience, 46% of the participants chose packet loss as the primary factor that affects the quality of real-time audio and video communication.(1)
Delay jitter in a speech signal may have various causes. The most frequent cause is the dynamic adaptation of the jitter buffers built into modern VoLTE applications. The longer these buffers are, the more packet jitter they may compensate for, but the latency of the speech signal is also increased. On the other hand, if the jitter buffer is shorter, the latency is also shorter, but the danger of packet loss is significantly higher.
The advantages of the EVS Audio Codec
EVS is currently the highest quality voice codec available for mobile voice communication. The EVS codec is designed for VoLTE/RCS and it enables Full-HD Voice with the best audio quality score in the market today for music and mixed content in conversational applications. EVS offers a wide range of bitrates to improve system efficiency through lower average bitrate and allows service providers to optimize network and call quality. In addition, some EVS modes use unique packet loss concealment and Jitter Buffer Management (JBM), which minimize degradation of voice quality.
With these and other advantages optimized for packet-based communications, developers and operators are spurred to bring implementations of the new EVS codec to market rapidly in an effort to boost the quality of experience (QoE) for voice services. However, comprehensive testing of the broad EVS feature set throughout the device lifecycle can prove to be quite complex.
Measuring the effects of encoder, decoder and network conditions, such as packet loss, latency and delay, on audio quality is challenging. Currently there are no agreed thresholds in any of the relevant standard bodies (e.g., ITU-T, IETF).
EVS testing and measurement is complex because there are many codec configurations, features, parameters, and audio quality measurements to take into consideration including:
Multi-rate audio codec
Voice/sound activity detector (VAD)
Comfort noise generation (CNG) system
Error concealment (EC) mechanism to neutralize the detrimental effects of transmission errors resulting in lost packets
Channel-aware mode which further improves frame/packet error resilience
Jitter buffer management (JBM) to reduce the impact of variable transmission time
Super-wideband and full band audio bandwidths
Enhanced interoperation with the AMR-WB codec over all nine operational bitrates
Spirent provides a complete EVS test and measurement solution for the R&D market based on the latest 3GPP standard including call generation and measurement.
To get an idea of just how much difference an EVS codec can make, take a look (and listen) at the Public Safety Communications Research Speech Intelligibility Demo page.
1. Webinar: Voice and Video over IP Communications: Assessing and Improving User Experience, Radvision, June 2010