No audio on SRTP/ZRTP

Hi, I have achieved a functional Freeswitch configuration in that I can call both Freeswitch hosted services (e.g. talking clock) and make outbound VoIP calls through upstream provider. However this is only achievable using Linphone in Transport TLS + Encryption None config. The moment I switch Linphone Encryption to TLS or ZRTP, I get complete loss of audio (for both Freeswitch hosted and outbound VoIP). I have tried expanding the list of ciphers in "sip_tls_ciphers" and "rtp_sdes_suites" (both in vars.xml) but that makes no difference. Evidently SSL certs are correctly setup because otherwise Transport TLS would not work. I'm not quite sure where to go from here ? Laura