Hello. I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines: m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both? Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600) SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/> Please, advise. Thanks!