asterisk users - Jul 2014 - no audio on call from sipML5 in browsers to Asterisk 11 with DTLS-SRTP

Hi all !

I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in 
order to test WebRTC setup on my Asterisk PBX. I am using latest SVN 
version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677)

If I make calls from softphones (Zoiper, X-Lite), which do not support 
DTLS at all, I can hear the Echo Test sound.

BUT when I call from browser (I've tried latest Mozilla Firefox and 
Chrome on both Windows 8.1 x86 and Android 4.3), I can successfully 
establish a call but I can hear NO SOUND.

As it turns out (according to wireshark logs), Asterick is sending me 
DTLS handshake, on which nobody replies on my machine. And probably 
because of this, all the RTP traffic which goes afterwards, is not 
understood by my browser (because it?s encrypted I guess)

Am I missing something critical in order for this setup to function 
properly ?

I here by attach Asterisk console log dumps for a typical call 
(sipml5_ff_desktop_from_public_ip_hw_failure.log),and also a Firefox 
sipML5 console log (sipML_console.log).

Setup for sipML Live demo:

Display Name: ?sipML Firefox desktop?

Private Identity^* : 21111

Public Identity^* :sip:21111 at asterisk.server.dnsname:5062

Realm^* :asterisk.server.dnsname

Expert mode settings:

WebSocket Server URL: ws://devsrv1.clickphone.net:8088/ws

ICE Servers: [{ url: 'stun:stun.l.google.com:19302'}]

This is a dump of DTLS packet that my machine receives but does not answer:

528334.683910000asterisk-ipmy-public-ipDTLSv1.0216Client Hello1544852833

000000 17 31 4c 2a 3f 00 21 91 c7 d8 89 08 00 45 00..1L*?.!......E.

001000 ca ed 57 40 00 35 11 be 64 90 4c c6 db 6d cb...W at .5..d.L..m.

0020d4 73 3c 58 ce 61 00 b6 f2 6a 16 fe ff 00 00 00.s<X.a...j......

003000 00 00 00 00 00 a1 01 00 00 95 00 00 00 00 00................

004000 00 95 fe ff 53 ba 93 1b bc 58 87 53 38 9e c9.....S....X.S8..

005015 ba bc 0a 5b 4d a5 0b 98 e9 e9 bd 8a 1d f8 c3....[M..........

006096 98 f3 b9 50 00 00 00 58 c0 14 c0 0a c0 22 c0....P...X.....".

007021 00 39 00 38 00 88 00 87 c0 0f c0 05 00 35 00!.9.8.........5.

008084 c0 12 c0 08 c0 1c c0 1b 00 16 00 13 c0 0d c0................

009003 00 0a c0 13 c0 09 c0 1f c0 1e 00 33 00 32 00............3.2.

00a09a 00 99 00 45 00 44 c0 0e c0 04 00 2f 00 96 00....E.D...../...

00b041 00 15 00 12 00 09 00 14 00 11 00 08 00 06 00A...............

00c0ff 02 01 00 00 12 00 23 00 00 00 0f 00 01 01 00.......#........

00d00e 00 05 00 02 00 01 00........

Setup of Asterisk PBX is described in the attachment asterisk_config.txt

Any help is HIGHLY appreciated !!

I wonder if anyone has ever really made calls from DTLS-SRTP 
WebRTC-client to Asterisk..

Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140707/1e806c9b/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: 2send.zip
Type: application/x-zip-compressed
Size: 6963 bytes
Desc: not available
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140707/1e806c9b/attachment.bin>