Performance comparison of various VoIP codecs in wireless environments
Kyungtae Kim, Young‐June Choi · 2011
One of the major challenges facing VoIP deployment is achieving and maintaining acceptable voice quality. The user-perceivable voice quality within a VoIP network has to match that of conventional circuit-based networks. This perceived voice quality is affected by many factors including delay, jitter, packet loss and used voice codec systems. Most of these factors depend on the bandwidth capacity and other characteristics of the network links. Thus, network monitoring/management techniques covering bandwidth, jitter, delay and packet loss should be employed to properly handle VoIP traffic. In addition, since call loads vary significantly with the sampling rates provided by different codec systems, such as G.711, G.729 and iLBC, correct selection and usage of a VoIP codec system are crucial to voice quality. In this paper, performances of different codecs such as G.711, iLBC, GSM, Speex, and Skype's codec are compared over the real wireless mesh network with different network conditions. Test results provide estimation on the supportable number of VoIP calls over the wireless mesh network, and useful information on each codec, such as robustness on the delay and packet drops.