Chrome m35(forces DTLS by default) <-> asterisk 11.9 [fix]

[GoogleCodeExporter]

1. Setup Asterisk using 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
2. Call any peer from sipml5

What is the expected output? What do you see instead?
It should start ringing, but fails with 'recv=SIP/2.0 488 Not acceptable here' 
on callee side.
Cause: https://code.google.com/p/webrtc/issues/detail?id=2774 (default changed 
from  SDES to DTLS)

What version of the product are you using? On what operating system?
Chrome M35, AsteriskNOW 11.9

Can be fixed in SIPml-api.js?svn=224:1063 by adding
'optional': [{DtlsSrtpKeyAgreement: false}] to o_media_constraints property in 
tmedia_session_jsep01 constructor.
To test it just add attached js monkeypatch as <script> after SIPml-api.js

SEND: INVITE sip:355@192.168.0.134 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bK7UnPGnzKF3lrrTNw9y85qP691hfS1ZND;rport
From: "354"<sip:354@192.168.0.134>;tag=79sUCJJTTPKhXUhtPV1C
To: <sip:355@192.168.0.134>
Contact: 
"354"<sip:354@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2d554a36-be07-326f-1deb-095678479b34
CSeq: 15065 INVITE
Content-Type: application/sdp
Content-Length: 1527
Max-Forwards: 70
Authorization: Digest 
username="354",realm="asterisk",nonce="02e21bdd",uri="sip:355@192.168.0.134",res
ponse="04fe8b4eb70f048c94c772d131f57775",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 3280650824429877000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS pNfv6i1IYt5ogoS5oOluhcqPRPFDonBTyy1B
m=audio 56963 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.0.139
a=rtcp:56963 IN IP4 192.168.0.139
a=candidate:1168604572 1 udp 2122260223 192.168.0.139 56963 typ host generation 
0
a=candidate:1168604572 2 udp 2122260223 192.168.0.139 56963 typ host generation 
0
a=candidate:186941804 1 tcp 1518280447 192.168.0.139 0 typ host generation 0
a=candidate:186941804 2 tcp 1518280447 192.168.0.139 0 typ host generation 0
a=ice-ufrag:fH81hLcZUKevaS/0
a=ice-pwd:YD5sE75egze5BXJpbDypVJDG
a=ice-options:google-ice
a=fingerprint:sha-256 
7C:CC:CF:72:62:70:5C:DB:73:1E:4A:A8:E3:21:26:9B:DD:9E:B3:85:AB:69:3B:10:5D:58:E2
:3A:FF:FC:C5:4E
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-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:568380636 cname:NoVQztPveQAiwz1o
a=ssrc:568380636 msid:pNfv6i1IYt5ogoS5oOluhcqPRPFDonBTyy1B 
90f7061a-4919-45fd-aeed-51cc2cffdbd0
a=ssrc:568380636 mslabel:pNfv6i1IYt5ogoS5oOluhcqPRPFDonBTyy1B
a=ssrc:568380636 label:90f7061a-4919-45fd-aeed-51cc2cffdbd0
 SIPml-api.js?svn=224:1063

recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=192.168.0.139;branch=z9hG4bK7UnPGnzKF3lrrTNw
9y85qP691hfS1ZND
From: "354"<sip:354@192.168.0.134>;tag=79sUCJJTTPKhXUhtPV1C
To: <sip:355@192.168.0.134>;tag=as1a6d11ba
Call-ID: 2d554a36-be07-326f-1deb-095678479b34
CSeq: 15065 INVITE
Content-Length: 0
Server: Asterisk PBX 11.9.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

Original issue reported on code.google.com by Each.nir...@gmail.com on 2 Jun 2014 at 8:39

Attachments:

• chrome35-asterisk_dtls-sdes_fix.js

[GoogleCodeExporter]

DtlsSrtpKeyAgreement will be removed in the next Chrome major version. DTLS is 
mandatory and you should start learning to live with it.

Original comment by boss...@yahoo.fr on 2 Jun 2014 at 2:17

[GoogleCodeExporter]

>>you should start learning to live with 
How? :) 
This unexpected 'mandatory' breaks working solutions. Proposed fix is fastest 
way to make things working again.   

Original comment by Each.nir...@gmail.com on 3 Jun 2014 at 7:56

[GoogleCodeExporter]

Try this patch:
https://issues.asterisk.org/jira/browse/ASTERISK-22961
works fine for me with Asterisk 11.10.2 on Ubuntu Trusty

Original comment by virmanta...@lamoda.ru on 19 Jun 2014 at 7:04

[GoogleCodeExporter]

which patch did you apply exactly? none of the patches work expect if i svn ast 
11.6 which finally works with webrtc but has one way audio

Original comment by bhakim...@gmail.com on 7 Jul 2014 at 2:19

[GoogleCodeExporter]

Check out latest 11 brach from svn. All working patches were merged there.

Original comment by virmanta...@gmail.com on 7 Jul 2014 at 4:50

[GoogleCodeExporter]

This issue also breaks communication with FreeSWITCH (Version 
1.5.13b+git~20140519T234223Z~bf84e9d414~64bit) in Chrome 37. Any workarounds?

Original comment by ksax...@5shells.com on 27 Aug 2014 at 5:26