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