An Evaluation of Secure Real-Time Transport Protocol (SRTP) Performance for VoIP

The Secure Real-Time Transport Protocol (SRTP) is an Internet standards-track security profile for RTP used to provide confidentiality, integrity and replay protection for RTP traffic. We study the performance of SRTP when it is used to secure VoIP conversations. Experiments are conducted using snom and Twinkle softphones running on Windows and Linux platforms respectively and a bare PC softphone running with no operating system installed to provide a baseline. Pre-defined SRTP transforms based on AES counter mode encryption with a 128-bit key and HMAC-SHA-1 with a 32-bit authentication tag, as well as 192 and 256-bit AES keys and an 80-bit authentication tag are tested. Measurement of internal processing times for each operation in the SRTP protocol indicates that authentication processing is more expensive than encryption regardless of key or tag size. A comparison of jitter and delta (packet interarrival time) for secured and unsecured VoIP traffic reveals that the addition of SRTP protection to VoIP traffic over RTP has a negligible effect on voice quality. VoIP throughput with SRTP is about 2% more than with RTP alone since the insignificant increase in delay is offset by the small increase in packet size.