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Abstract-Many mobile handsets manufactured today are equipped with wireless data modules that can provide users with the alternative to access Internet telephony service for voice communications. Due to the limited coverage area of wireless data service, however, a call established through VoIP service may need to be transferred transparently to mobile telephony service, and vice versa, for maintaining voice call continuity. In this paper, we investigate the problem of supporting seamless voice communications across heterogeneous telephony systems on dual-mode mobile devices such as GSM-Wi-Fi handsets. While related work has investigated the problem of seamless vertical handoffs across heterogeneous wireless data networks, solutions based on packet-switched protocols cannot be used in this context since GSM is a circuit-switched telephony system. Rather, to enable seamless voice communications across heterogeneous telephony systems, we show in this paper that the support of digital signal processing (DSP) techniques during handoffs is critical. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for supporting vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless voice call continuity across circuit-switched and packet-switched systems and explain why a "make-before-break" handoff with DSP support is necessary. We thus propose a solution that incorporates time scaling algorithms to process voice streams during handoffs for supporting seamless voice call continuity, and we investigate mechanisms to reduce the overheads of the proposed solution. To evaluate the performance of the proposed solution, we conduct testbed experiments using a GSM-Wi-Fi dual-mode notebook. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.

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As IP telephony gains more popularity, interworking with conventional PSTN telephony has also gained more importance. In particular, an increasing number of new telephony services now involves both packet-switched (IP telephony) and circuit-switched (PSTN telephony) voice legs in one call session. One common problem that arises for enabling such new services is the need for synchronization of voice streams that traverse through heterogeneous telephony systems. In this paper, we first identify the key role of voice synchronization across heterogeneous telephony systems for services such as seamless handover between WLAN and cellular networks and multi-party audio conferencing with video overlay. We then explain the challenges in synchronizing circuit-switched and packet-switched voice streams, including codec distortion, packet losses, line noises, and overlapping utterances. To achieve voice synchronization, we proceed to investigate three different approaches based on digital speech processing techniques in the waveform, cepstrum, and spectrum domains. Finally, we compare the performance benefits and tradeoffs of different approaches, thus motivating further research along this direction.