Getting - SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.

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b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1.Using Last official version of Asterisk 11.7.0
2.Any time making the call from Chrome or FireFox to another Chrome or FireFox 
browser and or SIP endpoint 
3.Getting the same error all the time

What is the expected output? What do you see instead?
Error in the browser

SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. 
SIPml-api.js?svn=179:1
tsk_utils_log_error SIPml-api.js?svn=179:1
tmedia_session_jsep01.onSetRemoteDescriptionError SIPml-api.js?svn=179:3
(anonymous function)


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

Asterisk 11.7.0 and Last version of SIPml-appi.js


Please provide any additional information below.

We are getting the following error in Asterisk
Reason: SIP; cause=603; text="Failed to get local SDP"


<--- SIP read from WS:xxx.xxx.xxx.75:52281 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS xxx.xxx.xxx.63:5060;rport=5060;branch=z9hG4bK4b059bcd
From: "1061"<sip:1061@xxx.xxx.xxx.63>;tag=as4d3911e5
To: 
<sip:1065@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=RgGtt7QdiqOK
ZWzc0KIp
Call-ID: 53a2c62a7358a49e5e07464c2a88db58@xxx.xxx.xxx.63:5060
CSeq: 102 INVITE
Content-Length: 0

==========================================================
Reason: SIP; cause=603; text="Failed to get local SDP"

===========================================================

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:1065@df7jal23ls0d.invalid;transport=ws> for 
address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to xxx.xxx.xxx.75:52281:
ACK sip:1065@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS xxx.xxx.xxx.63:5060;branch=z9hG4bK4b059bcd;rport
Max-Forwards: 70
From: "1061" <sip:1061@xxx.xxx.xxx.63>;tag=as4d3911e5
To: 
<sip:1065@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=RgGtt7QdiqOK
ZWzc0KIp
Contact: <sip:1061@xxx.xxx.xxx.63:5060;transport=WS>
Call-ID: 53a2c62a7358a49e5e07464c2a88db58@xxx.xxx.xxx.63:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0


---

<--- Reliably Transmitting (NAT) to 10.110.1.254:9508 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 
10.110.1.254:9508;branch=z9hG4bK-d8754z-e9a10870a18f842b-1---d8754z-;received=10
.110.1.254;rport=9508
From: "1061"<sip:1061@xxx.xxx.xxx.63>;tag=c3729302
To: <sip:1065@204.174.104.63>;tag=as6a707ef2
Call-ID: MzA2MGIwZGU0NzQ0MzgxNDZkNWM0ZWI0YjdiNDRiYzc
CSeq: 2 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

Original issue reported on code.google.com by aram...@gmail.com on 5 Feb 2014 at 8:58

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[deleted comment]

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You must enable ICE on the remote party

Original comment by boss...@yahoo.fr on 13 Dec 2014 at 6:50

• Changed state: Done