[WebRTC][Warning Rejecting secure audio stream without encr]

Hi All,
I am getting this while having call through
sipml5.org/expert.html

WARNING[14660][C-00000001]: chan_sip.c:10475 process_sdp: Rejecting secure audio stream without encryption details: audio 56308 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126

Help needed
Regards
Bilal Abbasi

This is my SIP debug Logs.
<— SIP read from WS:10.114.11.39:60482 —>
INVITE sip:888@10.114.5.79 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKTmlKdG4gOmsXKhNbvywLACwKwZPVFxf;rport
From: "2222"sip:2222@10.114.5.79;tag=CSZWRPDBcZi6g8GdU61J
To: sip:888@10.114.5.79
Contact: "2222"sip:2222@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language="en,fr"
Call-ID: 27719df7-197f-6fa7-b89d-0bdcd0c2618c
CSeq: 3427 INVITE
Content-Type: application/sdp
Content-Length: 2609
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 8891431215450093000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 8KZULc51oS3CYSb5d6ndxVwf3nK1Xbaon2K2
m=audio 59368 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 10.114.11.39
a=rtcp:59368 IN IP4 10.114.11.39
a=candidate:1947334831 1 udp 2122260223 10.114.11.39 59368 typ host generation 0
a=candidate:1947334831 2 udp 2122260223 10.114.11.39 59368 typ host generation 0
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 59369 typ host generation 0
a=candidate:2999745851 2 udp 2122194687 192.168.56.1 59369 typ host generation 0
a=candidate:3242075715 1 udp 2122129151 192.168.147.1 59370 typ host generation 0
a=candidate:3242075715 2 udp 2122129151 192.168.147.1 59370 typ host generation 0
a=candidate:2364966728 1 udp 2122063615 192.168.145.1 59371 typ host generation 0
a=candidate:2364966728 2 udp 2122063615 192.168.145.1 59371 typ host generation 0
a=candidate:982647903 1 tcp 1518280447 10.114.11.39 0 typ host tcptype active generation 0
a=candidate:982647903 2 tcp 1518280447 10.114.11.39 0 typ host tcptype active generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
a=candidate:2411584179 1 tcp 1518149375 192.168.147.1 0 typ host tcptype active generation 0
a=candidate:2411584179 2 tcp 1518149375 192.168.147.1 0 typ host tcptype active generation 0
a=candidate:3262479288 1 tcp 1518083839 192.168.145.1 0 typ host tcptype active generation 0
a=candidate:3262479288 2 tcp 1518083839 192.168.145.1 0 typ host tcptype active generation 0
a=ice-ufrag:rtPFv+pWvloeb/xZ
a=ice-pwd:QyluBPb0lzpGH1lpF83RsuX/
a=ice-options:google-ice
a=fingerprint:sha-256 10:37:A1:33:E5:FD:EE:94:9E:2C:7D:D8:A7:62:F6:22:49:9B:5F:8A:DD:73:FE:79:F8:DB:97:06:18:6C:5C:35
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 webrtc.org/experiments/rtp-h … -send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:801361830 cname:PyBMLRfbyK/qISET
a=ssrc:801361830 msid:8KZULc51oS3CYSb5d6ndxVwf3nK1Xbaon2K2 b6c40efc-244b-4786-805a-4efd64e5516d
a=ssrc:801361830 mslabel:8KZULc51oS3CYSb5d6ndxVwf3nK1Xbaon2K2
a=ssrc:801361830 label:b6c40efc-244b-4786-805a-4efd64e5516d
<------------->
— (12 headers 50 lines) —
Using INVITE request as basis request - 27719df7-197f-6fa7-b89d-0bdcd0c2618c
Found peer ‘2222’ for ‘2222’ from 10.114.11.39:60482
Retransmitting #3 (NAT) to 10.114.5.61:5080:
OPTIONS sip:10.114.5.61 SIP/2.0
Via: SIP/2.0/UDP 10.114.5.79:5060;branch=z9hG4bK01ab19ef;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.114.5.79;tag=as15b5b50b
To: sip:10.114.5.61
Contact: sip:asterisk@10.114.5.79:5060
Call-ID: 60d8930f2edb401104fc8d412c18c875@10.114.5.79:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-trunk-r379070M
Date: Fri, 02 Jan 2015 15:41:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Reliably Transmitting (NAT) to 10.114.11.39:60482:
OPTIONS sip:2222@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.114.5.79:5060;branch=z9hG4bK192f5042;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.114.5.79;tag=as350eca4f
To: sip:2222@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
Contact: sip:asterisk@10.114.5.79:5060;transport=WS
Call-ID: 6d7c4a5e46c861dc1aef15a81b50a09c@10.114.5.79:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX SVN-trunk-r379070M
Date: Fri, 02 Jan 2015 15:41:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
[Jan 2 20:41:36] WARNING[14660][C-00000002]: chan_sip.c:10475 process_sdp: Rejecting secure audio stream without encryption details: audio 59368 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126

<— Reliably Transmitting (NAT) to 10.114.11.39:60482 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKTmlKdG4gOmsXKhNbvywLACwKwZPVFxf;received=10.114.11.39;rport=60482
From: "2222"sip:2222@10.114.5.79;tag=CSZWRPDBcZi6g8GdU61J
To: sip:888@10.114.5.79;tag=as426737e1
Call-ID: 27719df7-197f-6fa7-b89d-0bdcd0c2618c
CSeq: 3427 INVITE
Server: Asterisk PBX SVN-trunk-r379070M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘27719df7-197f-6fa7-b89d-0bdcd0c2618c’ in 13056 ms (Method: INVITE)

<— SIP read from WS:10.114.11.39:60482 —>
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 10.114.5.79:5060;rport=5060;branch=z9hG4bK192f5042
From: "asterisk"sip:asterisk@10.114.5.79;tag=as350eca4f
To: sip:2222@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
Call-ID: 6d7c4a5e46c861dc1aef15a81b50a09c@10.114.5.79:5060
CSeq: 102 OPTIONS
Content-Length: 0