No audio on SRTP/ZRTP

Rachel Roch rroch at
Mon Oct 19 12:41:44 UTC 2020


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 ?


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