AMR-WB is specified by 3GPP as the mandatory codec to be supported by wideband mobile terminals for a wide range of communication services. This is mentioned here.https://tools.ietf.org/html/draft-marjou-rtcweb-audio-codecs-for-interop-01#ref-AMR-WB
This includes the massively deployed circuit switched mobile telephony services and new multimedia telephony services over IP/IMS and 4G/VoLTE as specified by GSMA as voice IMS profile for VoLTE in IR92. Hence, AMR-WB is strongly increasing with deployment in more than 60 networks from 45 countries and more than 130 types of terminals. In that use case, if OPUS and G.711 remain the only codecs supported by the WebRTC endpoints, a gateway must then transcode these codecs into AMR-WB, and vice-versa, in order to implement the use-case. As a consequence, a high number of calls are likely to be affected by transcoding operations producing a degradation of the user quality experience for many customers. This will have a very significant business impact for all service providers on both sides, not only with respect to the transcoding costs but mainly with respect to user experience degradation.
The drawbacks of transcoding are recalled below:
Cost issues: transcoding places important additional costs on network gateways for example codec implementation and license costs, deployments costs, testing/validation costs etc... However these costs can be seen as just transferred from the terminal side to the network side. The real issue is rather the degradation of the quality of service affecting the end user perceived quality which will be harmful to all concerned service providers.
Intrinsic quality degradation: Subjective test results show that intrinsic voice quality is significantly degraded by transcoding. The degradation is around 0.2 to 0.3 MOS for most of transcoding use cases with AMR-WB at 12.65 kbit/s. It should be stressed that if transcoding is performed between AMR-WB and G.711, wideband voice quality will be lost. Such bandwidth reduction effect clearly degrades the user perceived quality of service leading to shorter and less frequent calls (see ref_gsma). Such a switch to G.711 will not be accepted anymore by customers. If transcoding is performed between AMR-WB and OPUS, wideband communication could be maintained. However, as the WB codecs complexity is higher than NB codecs complexity, such WB transcoding is also more costly and degrades the quality: MOS scores of transcoding between AMR-WB 12.65kbit/s and OPUS at 16 kbit/s in both directions are significantly lower than those of AMR-WB at 12.65kbit/s or OPUS at 16 kbit/s. Furthermore, in degraded conditions, the addition of defects, like audio artifacts due to packet losses, and the audio effects resulting from the cascading of different packet loss recovery algorithms may result in a quality below the acceptable limit for the customers.
Degraded interactivity due to increased latency: Transcoding means full de-packetization for decoding of the media stream (including mechanisms of de-jitter buffering and packet loss recovery) then re-encoding, re-packetization and re-sending. The delays produced by all these operations are additive and may increase the end to end delay beyond acceptable limits like with more than 1s end to end latency.
As a consequence, re-using AMR-WB would be beneficial for the specific usage of WebRTC technology over mobile networks. With the strong increase of the smartphone market the capability to use such a mobile codec could strongly enforce and extend the market penetration of the Web RTC technology.
references:
https://tools.ietf.org/html/draft-marjou-rtcweb-audio-codecs-for-interop-01#section-4.1
This includes the massively deployed circuit switched mobile telephony services and new multimedia telephony services over IP/IMS and 4G/VoLTE as specified by GSMA as voice IMS profile for VoLTE in IR92. Hence, AMR-WB is strongly increasing with deployment in more than 60 networks from 45 countries and more than 130 types of terminals. In that use case, if OPUS and G.711 remain the only codecs supported by the WebRTC endpoints, a gateway must then transcode these codecs into AMR-WB, and vice-versa, in order to implement the use-case. As a consequence, a high number of calls are likely to be affected by transcoding operations producing a degradation of the user quality experience for many customers. This will have a very significant business impact for all service providers on both sides, not only with respect to the transcoding costs but mainly with respect to user experience degradation.
The drawbacks of transcoding are recalled below:
Cost issues: transcoding places important additional costs on network gateways for example codec implementation and license costs, deployments costs, testing/validation costs etc... However these costs can be seen as just transferred from the terminal side to the network side. The real issue is rather the degradation of the quality of service affecting the end user perceived quality which will be harmful to all concerned service providers.
Intrinsic quality degradation: Subjective test results show that intrinsic voice quality is significantly degraded by transcoding. The degradation is around 0.2 to 0.3 MOS for most of transcoding use cases with AMR-WB at 12.65 kbit/s. It should be stressed that if transcoding is performed between AMR-WB and G.711, wideband voice quality will be lost. Such bandwidth reduction effect clearly degrades the user perceived quality of service leading to shorter and less frequent calls (see ref_gsma). Such a switch to G.711 will not be accepted anymore by customers. If transcoding is performed between AMR-WB and OPUS, wideband communication could be maintained. However, as the WB codecs complexity is higher than NB codecs complexity, such WB transcoding is also more costly and degrades the quality: MOS scores of transcoding between AMR-WB 12.65kbit/s and OPUS at 16 kbit/s in both directions are significantly lower than those of AMR-WB at 12.65kbit/s or OPUS at 16 kbit/s. Furthermore, in degraded conditions, the addition of defects, like audio artifacts due to packet losses, and the audio effects resulting from the cascading of different packet loss recovery algorithms may result in a quality below the acceptable limit for the customers.
Degraded interactivity due to increased latency: Transcoding means full de-packetization for decoding of the media stream (including mechanisms of de-jitter buffering and packet loss recovery) then re-encoding, re-packetization and re-sending. The delays produced by all these operations are additive and may increase the end to end delay beyond acceptable limits like with more than 1s end to end latency.
As a consequence, re-using AMR-WB would be beneficial for the specific usage of WebRTC technology over mobile networks. With the strong increase of the smartphone market the capability to use such a mobile codec could strongly enforce and extend the market penetration of the Web RTC technology.
references:
https://tools.ietf.org/html/draft-marjou-rtcweb-audio-codecs-for-interop-01#section-4.1
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