chrome to Asterisk call: SYNTAX_ERR: DOM Exception 12 upon 200 OK from Asterisk

Hello,

I followed the instructions from the following page:

http://code.google.com/p/sipml5/wiki/Asterisk

and I am able to establish a call from Google Chrome to an Asterisk server onto 
the same LAN, but the audio transmission in both directions does not seem to 
work at all.

The SYNTAX_ERR: DOM Exception 12 happens upon receiving the 200 OK response and 
I think it happens in native code called within this.o_pc.setRemoteDescription, 
which in turn is called from tmedia_session_jsep.prototype.__set_ro, so there's 
probably something wrong with the SDP received from Asterisk.

What steps will reproduce the problem?

1. Open ML5 page call.htm
2. Set the SIP account parameters and WebSocket server (I am using the 
repro-based one downloadable from your site, not Asterisk itself) 
3. Place a call

What is the expected output? What do you see instead?

I expect to be able to hear/send audio but instead it doesn't work.

What version of the product are you using? On what operating system?

Chrome 24.0.1297.0 dev-m on Win 7 Professional SP1 x64

Please provide any additional information below.

This is the Chrome-side log:

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:55
SEND: INVITE sip:90227070011@10.77.36.12 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKkDFyAEJ8yxyFs8XKzEzZPmaXbTbrGpTT;rport
From: <sip:8600@10.77.36.12>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:90227070011@10.77.36.12>
Contact: "fabry"<sip:8600@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Type: application/sdp
Content-Length: 1146
Max-Forwards: 70
Authorization: Digest 
username="8600",realm="asterisk",nonce="00112101",uri="sip:90227070011@10.77.36.
12",response="04fc599e2a496a1b71f881c1f67527b2",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 3617092930 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 60443 RTP/SAVP 103 104 0 8 106 105 13 126
c=IN IP4 10.77.35.15
a=rtcp:60443 IN IP4 10.77.35.15
a=candidate:1697405919 1 udp 2113937151 10.77.35.15 60443 typ host generation 0
a=candidate:1697405919 2 udp 2113937151 10.77.35.15 60443 typ host generation 0
a=candidate:732931887 1 tcp 1509957375 10.77.35.15 53043 typ host generation 0
a=candidate:732931887 2 tcp 1509957375 10.77.35.15 53043 typ host generation 0
a=ice-ufrag:lfzNvrLdcOa18IiY
a=ice-pwd:dhwmGn1IroVQKLxYBh8aR64N
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:Q4unJxQVjw+oh5PvWvk3UophXn2LQAPHL2F7A8MK
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=ssrc:3374699521 cname:k+3tRzqrO9GHIppN
a=ssrc:3374699521 mslabel:IDB9Uz1ISXFcDZhBc8ZktMV31LWwrQlaNpN4
a=ssrc:3374699521 label:IDB9Uz1ISXFcDZhBc8ZktMV31LWwrQlaNpN400
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyF
s8XKzEzZPmaXbTbrGpTT
From: <sip:8600@10.77.36.12>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:90227070011@10.77.36.12>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Length: 0

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Trying tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 
10.77.35.15:53036;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyFs8X
KzEzZPmaXbTbrGpTT
From: <sip:8600@10.77.36.12>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:90227070011@10.77.36.12>;tag=as028f7b42
Contact: <sip:90227070011@10.77.36.12:5060>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Type: application/sdp
Content-Length: 576
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Server: Asterisk PBX SVN-trunk-r373330M
Supported: replaces,timer

v=0
o=root 1547035746 1547035746 IN IP4 10.77.36.12
s=Asterisk PBX SVN-trunk-r373330M
c=IN IP4 10.77.36.12
t=0 0
m=audio 12778 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0a35d5ab2e4d42df3fc1262a56cbe223
a=ice-pwd:2786b50c4df31cbc3705554b5b6481a8
a=candidate:Ha4d240c 1 udp 2130706431 10.77.36.12 12778 typ host
a=candidate:Ha4d240c 2 udp 2130706430 10.77.36.12 12779 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:AsXbqqQscS5j6Fq044NwY6b/0HumDyWpPOz/VwtR
 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
__on_open tsk_utils.js:55
__on_state_change tsk_utils.js:55
2Session Progress tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 
10.77.35.15:53036;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyFs8X
KzEzZPmaXbTbrGpTT
From: <sip:8600@10.77.36.12>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:90227070011@10.77.36.12>;tag=as028f7b42
Contact: <sip:90227070011@10.77.36.12:5060>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Length: 0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Server: Asterisk PBX SVN-trunk-r373330M
Supported: replaces,timer

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Ringing tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 200 OK
Via: SIP/2.0/TCP 
10.77.35.15:53036;rport=53036;received=10.77.35.15;branch=z9hG4bKkDFyAEJ8yxyFs8X
KzEzZPmaXbTbrGpTT
From: <sip:8600@10.77.36.12>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:90227070011@10.77.36.12>;tag=as028f7b42
Contact: <sip:90227070011@10.77.36.12:5060>
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 INVITE
Content-Type: application/sdp
Content-Length: 576
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Server: Asterisk PBX SVN-trunk-r373330M
Supported: replaces,timer

v=0
o=root 1547035746 1547035747 IN IP4 10.77.36.12
s=Asterisk PBX SVN-trunk-r373330M
c=IN IP4 10.77.36.12
t=0 0
m=audio 12778 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0a35d5ab2e4d42df3fc1262a56cbe223
a=ice-pwd:2786b50c4df31cbc3705554b5b6481a8
a=candidate:Ha4d240c 1 udp 2130706431 10.77.36.12 12778 typ host
a=candidate:Ha4d240c 2 udp 2130706430 10.77.36.12 12779 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:AsXbqqQscS5j6Fq044NwY6b/0HumDyWpPOz/VwtR
 tsk_utils.js:55
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:55
DOMException {message: "SYNTAX_ERR: DOM Exception 12", name: "SYNTAX_ERR", 
code: 12, stack: "Error: An invalid or illegal string was 
specified.…c/transactions/tsip_transac_ict.js?svn=10:454:32)"}
 tsk_utils.js:67
SEND: ACK sip:90227070011@10.77.36.12:5060 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcDlNGy6AXElWlUWib27B;rport
From: <sip:8600@10.77.36.12>;tag=fTAsAMFb3z3fgPP4CIkJ
To: <sip:90227070011@10.77.36.12>;tag=as028f7b42
Contact: "fabry"<sip:8600@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 87d06336-ca33-f248-d7d6-a2ec7b07ea4f
CSeq: 31932 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="8600",realm="asterisk",nonce="00112101",uri="sip:90227070011@10.77.36.
12:5060",response="826f96616377356598a830f046abdc01",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

On the Asterisk side, there are no warnings or errors.
I noticed that the SDP returned by Asterisk has no a=rtcp line, could that be 
the problem?

Thanks in advance.
Best regards,
Fabrizio Ammollo
Original issue reported on code.google.com by f.ammo...@gmail.com on 18 Oct 2012 at 9:42
kamorahul / sipml5

chrome to Asterisk call: SYNTAX_ERR: DOM Exception 12 upon 200 OK from Asterisk #48
