No audio on SRTP/ZRTP
rroch at tutanota.de
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 ?
More information about the FreeSWITCH-users