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.

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the session initiation protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.

In this paper, we consider a university campus that has an established infrastructure for supporting SIP-based VoIP service through the campus wireless data network. The campus WLAN, however, does not have 100% full coverage, and hence users cannot make untethered VoIP calls anywhere on campus. The goal of this paper is to overcome the limitations of such WLAN dead spots and improve user experience when making VoIP calls. Different from related work that relies on peer-to-peer communication for multi-hop relay, we propose an approach called ldquodual-mode communicationrdquo to leverage the availability of dual-mode handsets for ubiquitous communication on campus. We implement the proposed approach and evaluate its performance in the campus testbed environment. We find that an opportunistic usage of dual communication modes on mobile handsets does allow ubiquitous voice communication in WLAN dead spots. However, it has one problem as the potential lack of voice call continuity during hand-off between the two modes that can cause degradation of the speech quality. We adopt a cross-layer solution based on signal processing algorithms to address the problem, thus achieving seamless voice call continuity while enabling ubiquitous voice communication on campus. Testbed evaluations show promising results for future research along the proposed direction.