Research on voice codec algorithms of SIP phone based on embedded system

Session Initiation Protocol (SIP) as a new multimedia communicating and instant messaging protocol drew more and more attentions recently. The software and hardware architecture of SIP phone which based on ARM920T core is demonstrated in this paper. The comparison of three voice codec algorithms including PCM, SPEEX and iLBC are implemented by porting these algorithms to embedded SIP phone platform. After several experiments, the result indicates that voice quality (e.g. MOS and R value) almost varies depending on the bandwidth, which also fit for theoretical analysis perfectly. Brings forward an effective conclusion that iLBC speech codec have excellent performance in low bit rates and it is superior to PCM and SPEEX encoding in abominable packet loss conditions. The experimental results also demonstrate that the SIP phone is suitable for voice communication and it can meet practical engineering requirements well.