Tomas Frankkila

Bandwidth efficient AMR operation for VoIP

An example of a bandwidth efficient adaptive multi rate (AMR) system for Voice over IP (VoIP) is presented. In VoIP, packet losses cause degradation of the synthesized speech. The distortions may propagate over several consecutive frames, since predictors in the codec exploit inter-frame correlations to gain coding efficiency. To reduce the effects of packet loss, forward error correction (FEC) that adds redundant information to voice packets can be used. However, while FEC can reduce the effects of packet loss, it will increase the amount of bandwidth used by the voice stream, which is not desirable. In this paper we propose FEC methods like partial redundancy, selective redundancy for the most sensitive frames and parameter interpolation in conjunction with AMR codec mode adaptation, which secure the speech quality when using AMR for VoIP without increasing the bandwidth substantially.

The cellular text telephone modem - the solution for supporting text telephone functionality in GSM networks

Text telephone devices are text-based terminals that allow the users to communicate by text via fixed-line telephone networks. Since cellular phone systems are sometime subject to severe radio channel impairments and the modem signals of these text telephones are therefore not always transmitted reliably, the Federal Communications Commission (FCC) has required a solution to guarantee a reliable transmission of text telephone data for emergency calls via cellular phone systems. For the North American PCS-1900 cellular phone systems the Standards Committee T1 has standardized a solution for this requirement, which is based on a new modem protocol, the cellular text telephone modem (CTM), whose signals can be reliably transmitted via the speech channel of cellular phone systems. After an introduction into text telephony, this contribution provides a description of this solution for PCS-1900 systems using CTM signals. The solution is indeed independent of the cellular system and works on de-facto all speech channels.

Realization and Performance Evaluation of IMS Multimedia Telephony for HSPA

The proposed voice over IP (VoIP) over HSPA solution targets high capacity as well as coverage and speech quality comparable to circuit switched (CS) speech. The VoIP over HSPA radio bearer realization includes one radio link control (RLC) un-acknowledged mode for the speech media. Parallel signaling radio bearers are also transmitted using HSPA access. Due to the flexibility of IP and HSPA, both narrow-band and wideband speech codecs work well. A delay sensitive scheduler is selected for the downlink transmission. For the uplink, the non-scheduled mode is selected. Coverage and quality can, with simulations, be shown to be at least as good as CS speech according to 3GPP ReI. 99 specifications. Similarly, capacity evaluations show that VoIP over HSPA has the potential of matching or exceeding CS speech capacity, depending on scenario. Finally, end to end simulations show that the maximum allowed delay of single links seldom occurs for uplink and downlink simultaneously, resulting in a lower perceived end-to-end delay than expected most of the time

System aspects of the 3GPP evolution towards enhanced voice services

The Enhanced Voice Services (EVS) codec was standardized by 3GPP in 2014. This codec offers significant gains in voice quality, efficiency, channel error robustness over any other existing speech codec and far better music quality. Operators run voice services on a large installed base of 3GPP Circuit-Switched (CS) 2G (GERAN) or 3G (UTRAN) radio networks. These networks offer mobile voice service either as HD voice using the AMR-WB codec, or as traditional narrowband (NB) voice service, based on the AMR codec. Voice over LTE (VoLTE) is currently being deployed throughout the world with HD Voice. The EVS codec will be first introduced as a straightforward VoLTE upgrade. It is also expected that EVS will be deployed over 3G CS networks. This paper describes system aspects relating to EVS codec introduction in VoLTE and CS networks as well as interworking and mobility with legacy systems and services.