Summary:	ASTERISK-23515: ICE info missing ice-ufrag and ice-pwd in INVITE received by caller
Reporter:	neeraj nagi (neerajnagi)	Labels:
Date Opened:	2014-03-22 07:09:47	Date Closed:	2014-04-08 12:18:14
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/General Channels/chan_sip/WebSocket Resources/res_http_websocket
Versions:	11.8.1	Frequency of Occurrence
Related Issues:	is related to ASTERISK-23425 No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.
Environment:	asterisk 11.8 on debian , with jssip client	Attachments:
Description:	Is there anyone using asterisk 11 successfully with jssip on latest chrome. I am having ice related issue, ice related info is missing in invite SDP exact issue in jssip console is on caller side SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. asterisk is not sending any ice information in sdp
Comments:	By: Rusty Newton (rnewton) 2014-03-25 12:46:04.432-0500 I believe your issue may be a duplicate of ASTERISK-23425 Can you provide more information about your configuration? You have told us what is happening and provided an error message, but you haven't provided any information on configuration or how to reproduce the issue. Please attach your sip.conf and rtp.conf files, the dialplan used, and show your configuration for the jssip client. Describe the process to reproduce the issue. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: neeraj nagi (neerajnagi) 2014-03-26 03:20:21.556-0500 I have tried three versions of asterisk 11.6 11.8 and 12. with 11.6 i am able to connect the call but one way audio.. caller voice is not through. with version 11.8 and 12 this is same as described in earlier mail. on the calle side JSSIP/sipml5 both show ice info missing. I can extend you access of the servers, if you want to debug the issue.. or tell me what files and logs i send you... thanks By: Rusty Newton (rnewton) 2014-03-31 17:21:38.298-0500 Please test with Asterisk 11.9.0-rc1 or above (or latest SVN) as we have a report that the issue no longer occurs in this version. By: Rusty Newton (rnewton) 2014-04-08 12:18:02.491-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines Please see ASTERISK-23425 for more information on the issue.