Integrating Secure RTP into the Open Source VoIP PBX Asterisk

Implementations of Voice over Internet Protocol (VoIP) have focused, up to now, mainly on the need to transport data in real-time, often at the expense of security. The neglect of secure VoIP is often intentional, as developers are striving to minimise overheads and delays. The Secure Real-Time Protocol (SRTP) has the potential to secure real-time streams without exacting too high a performance price. SRTP is the addition of security to the audio/video profile used in the Real-Time Transport Protocol (RTP). SRTP adds confidentiality, integrity and optionaly authenticity to RTP media streams. This paper focuses on the integration of SRTP into Asterisk, an open-source VoIP PBX. SRTP support has recently been added to Asterisk by Mikael Magnusson. This paper analyses Magnusson’s implementation, contrasting it to a proof-of-concept implementation developed independently at Rhodes University. The interoperability of SRTP implementations cannot be taken for granted, given the relatively recent standardization of the protocol, and so Magnusson’s implementation is tested against another SRTP implementation. Finally, the paper highlights a major shortcoming in Magnusson’s implementation, namely that the exchange of encryption keys is done in the clear. It concludes by proposing possible solutions, such as TLS, IPSec and MIkey.