Rejecting secure audio stream without encryption details

I’m playing with the sipml5 demo loaded on a local server. Using Chrome 40 I can connect to an account but cannot place a call. I’m trying to call locally from an extension on my freepbx distro server to another local extension. I’ve enabled AVPF and encrytion on both extension I’m testing with. Asterisk 13.1. Anyone have any ideas?

[2015-02-06 14:50:12] WARNING[11047][C-000152ee]: chan_sip.c:10503 process_sdp: Rejecting secure audio stream without encryption details: audio 19389 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
[2015-02-06 14:50:12] WARNING[11047][C-000152ee]: chan_sip.c:10503 process_sdp: Rejecting secure audio stream without encryption details: audio 19389 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126

Ok getting closer. Had to enable DTLS now I can place a call but cannot pick it up.

== Using SIP RTP CoS mark 5
-- Called SIP/1002
-- Connected line update to SIP/1001-0000ba5c prevented.
-- SIP/1002-0000ba5d is ringing
-- Got SIP response 603 "Failed to get local SDP" back from 10.0.0.88:52718
-- SIP/1002-0000ba5d is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [[email protected]:45] ExecIf("SIP/1001-0000ba5c", "0?MacroExit()") in new stack
[2015-02-06 16:18:42] WARNING[20006][C-000153d6]: chan_sip.c:24214 handle_response: Remote host can't  match    request ACK to call 'deleted'. Giving up.
[2015-02-06 16:18:42] WARNING[20006][C-000153d6]: chan_sip.c:24214 handle_response: Remote host can't match request ACK to call 'deleted'. Giving up.
-- Executing [[email protected]:46] ExecIf("SIP/1001-0000ba5c", "0?Set(DIALSTATUS=)") in new stack
-- Executing [[email protected]:47] GosubIf("SIP/1001-0000ba5c", "0?s-BUSY,1()") in new stack
-- Executing [[email protected]:48] MacroExit("SIP/1001-0000ba5c", "") in new stack
-- Executing [[email protected]:17] Set("SIP/1001-0000ba5c", "SV_DIALSTATUS=BUSY") in new stack
-- Executing [[email protected]:18] GosubIf("SIP/1001-0000ba5c", "0?docfu,1()") in new stack
-- Executing [[email protected]:19] GosubIf("SIP/1001-0000ba5c", "0?docfb,1()") in new stack
-- Executing [[email protected]:20] Set("SIP/1001-0000ba5c", "DIALSTATUS=BUSY") in new stack
-- Executing [[email protected]:21] ExecIf("SIP/1001-0000ba5c", "0?MacroExit()") in new stack
-- Executing [[email protected]:22] GotoIf("SIP/1001-0000ba5c", "1?s-BUSY,1") in new stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing [[email protected]:1] GotoIf("SIP/1001-0000ba5c", "0?exit,1") in new stack
-- Executing [[email protected]:2] PlayTones("SIP/1001-0000ba5c", "busy") in new stack
-- Executing [[email protected]:3] Busy("SIP/1001-0000ba5c", "20") in new stack
== Spawn extension (macro-exten-vm, s-BUSY, 3) exited non-zero on 'SIP/1001-0000ba5c' in macro 'exten-vm'
== Spawn extension (from-internal, 1002, 2) exited non-zero on 'SIP/1001-0000ba5c'
-- Executing [[email protected]:1] Hangup("SIP/1001-0000ba5c", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0000ba5c'

Honestly don’t waste you time with SIPML5. You’d have much better luck with sip.js or onsip.js