Closed GoogleCodeExporter closed 9 years ago
How can i fix this error ?
Original comment by jaballah...@gmail.com
on 27 Nov 2012 at 8:00
[deleted comment]
Anything new int this issue?
Original comment by Jakub.Tr...@gmail.com
on 2 Dec 2012 at 10:27
No, i still have this issue ...
Original comment by jaballah...@gmail.com
on 3 Dec 2012 at 7:29
I was using mac, and faced same issue. but on Win7 its working well.
Original comment by sad.bird...@gmail.com
on 3 Dec 2012 at 8:03
Yes, have this issue on Mac too
Original comment by Jakub.Tr...@gmail.com
on 3 Dec 2012 at 8:44
i tested with windows 7, windows 8 and Ubuntu. i have this issue when i use a
ovh sip
Original comment by jaballah...@gmail.com
on 3 Dec 2012 at 9:01
I see the same issue with an iptel.org sip. It gives the same error using
windows 8 and windows 7. My browser is Chrome.
Original comment by akxte...@gmail.com
on 20 Dec 2012 at 1:47
Fixed in API v1.1.0 (SVN r148)
http://code.google.com/p/sipml5/wiki/Downloads
Original comment by boss...@yahoo.fr
on 20 Dec 2012 at 7:13
it is not working, now with demo i got this error :
SIPml-api.js:1
PeerConnectionClass = function RTCPeerConnection() { [native code] }
SessionDescriptionClass = function RTCSessionDescription() { [native code] }
IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js:1
setRemoteDescription(offer) SIPml-api.js:1
SetRemoteDescription failed. SIPml-api.js:1
tsk_utils_log_error SIPml-api.js:1
(anonymous function) SIPml-api.js:3
tmedia_session_jsep01.__set_ro SIPml-api.js:3
tmedia_session_jsep01.__get_lo SIPml-api.js:3
tmedia_session.get_lo SIPml-api.js:1
tmedia_session_mgr.get_lo SIPml-api.js:1
tmedia_session_mgr.set_ro SIPml-api.js:1
tsip_dialog_invite.process_ro SIPml-api.js:3
__tsip_dialog_invite_cond_is_bad_content SIPml-api.js:3
tsk_fsm.act SIPml-api.js:1
tsip_dialog.fsm_act SIPml-api.js:3
__tsip_dialog_invite_event_callback SIPml-api.js:3
tsip_dialog.callback SIPml-api.js:3
__tsip_transac_ist_Started_2_Proceeding_X_INVITE SIPml-api.js:3
tsk_fsm.act SIPml-api.js:1
tsip_transac.fsm_act SIPml-api.js:3
tsip_transac_ist.start SIPml-api.js:3
tsip_dialog_layer.handle_incoming_message SIPml-api.js:3
tsip_transport_layer.handle_incoming_message SIPml-api.js:3
__tsip_transport_ws_onmessage SIPml-api.js:3
createAnswer SIPml-api.js:1
onCreateSdpError SIPml-api.js:1
State machine: tsip_dialog_invite_Started_2_Started_X_any SIPml-api.js:1
CreateAnswer can't be called before SetRemoteDescription. SIPml-api.js:1
tsk_utils_log_error SIPml-api.js:1
tmedia_session_jsep01.onCreateSdpError SIPml-api.js:3
(anonymous function) SIPml-api.js:3
tmedia_session_jsep01.__get_lo SIPml-api.js:3
tmedia_session.get_lo SIPml-api.js:1
tmedia_session_mgr.get_lo SIPml-api.js:1
tmedia_session_mgr.set_ro SIPml-api.js:1
tsip_dialog_invite.process_ro SIPml-api.js:3
__tsip_dialog_invite_cond_is_bad_content SIPml-api.js:3
tsk_fsm.act SIPml-api.js:1
tsip_dialog.fsm_act SIPml-api.js:3
__tsip_dialog_invite_event_callback SIPml-api.js:3
tsip_dialog.callback SIPml-api.js:3
__tsip_transac_ist_Started_2_Proceeding_X_INVITE SIPml-api.js:3
tsk_fsm.act SIPml-api.js:1
tsip_transac.fsm_act SIPml-api.js:3
tsip_transac_ist.start SIPml-api.js:3
tsip_dialog_layer.handle_incoming_message SIPml-api.js:3
tsip_transport_layer.handle_incoming_message SIPml-api.js:3
__tsip_transport_ws_onmessage SIPml-api.js:3
State machine: s0000_Started_2_Ringing_X_iINVITE
Original comment by jaballah...@gmail.com
on 20 Dec 2012 at 3:12
@jaballahrabie
You have to attach full log (from browser start to the error)
Original comment by boss...@yahoo.fr
on 20 Dec 2012 at 10:25
After the fix, when using the demo, when I call the iptel sip, sipml5 rings and
when I click receive, it stops ringing and instead of showing "failed to get
local SDP offer" it doesn't show an error at all.
However, the connection isn't actually made and the phone keeps on ringing on
the other end of the line.
Original comment by akxte...@gmail.com
on 28 Dec 2012 at 6:18
attaching javascript console log.
Original comment by akxte...@gmail.com
on 28 Dec 2012 at 6:27
Attachments:
Windows 7 64 Chrome, I see this bug too (
Original comment by 4587...@gmail.com
on 31 Dec 2012 at 8:05
@akxtech2
It's normal to get this error as your SDP do not contains some mandatory
features (SRTP, ICE...). Please enable RTCWeb Breaker
(http://sipml5.org/expert.htm) to fix the issue.
Original comment by boss...@yahoo.fr
on 1 Jan 2013 at 12:29
Hello,
I am getting same problem of akxtech2 that is when I am making call from web
browser to xlite client its showing the "failed to get local SDP offer"
1. SIP server I am using is - Officesip.
2. softphone - Xlite 3.0
I have already enabled the RTCWeb Browser
Original comment by saurabh4...@gmail.com
on 15 Mar 2013 at 9:58
@saurabh4u.hbti
I guess you meant "RTCWeb Breaker". Enabling 'RTCWeb Breaker' with officesip is
useless, you need webrtc2sip (http://webrtc2sip.org/).
officesip cannot connect chrome with xlite as it's not a media gateway
Original comment by boss...@yahoo.fr
on 15 Mar 2013 at 1:26
Am using window 7 64bit and chrome Version 29.0.1547.66 m and Asterisk version
11.1.2.
When I try to make a call I get, Got SIP response 603 "Failed to get local SDP.
Regards,
Original comment by sai...@gmail.com
on 13 Sep 2013 at 1:32
Just to help other people who might still run across this issue, I solved this
issue with inbound calls by disabling all video codecs for the Asterisk peer
that is using sipML5 (Also note in our case we don't allow Asterisk to reinvite
calls). It seems that in Chrome if H264 (and possibly other video codecs) is
set for the SIP peer the SDP includes it in the inbound offer it results in the
"failed to get local SDP" error. Oddly enough Firefox however did not have this
issue (Perhaps because it supports H264 unlike Chrome?). I didn't bother to
test VP8.
As of right now, I have gotten sipML5 audio calls to work perfectly with Chrome
and Firefox directly connecting to Asterisk 11.5.0 (Without WebRTCBreaker and
WebRTC4All), other than the Firefox hold/resume issue.
Original comment by CraigShe...@gmail.com
on 4 Feb 2015 at 9:36
Hi, we tested asterisk 13.2, asterisk 11.11, asterisk 12.8.1 wiith pjsip,
without pjsip, and we got the same issue about the SDP.
Sipml5 error on Chrome:
Failed to set remote offer sdp: Called with SDP without ice-ufrag and ice-pwd
Sipml5 error on Firefox:
ICE attributes missing; cause = MISSING_ICE_ATTRIBUTES
Asterisk Error:
SIP/w2417-00000005 is ringing
-- Got SIP response 603 "Failed to get local SDP" back from
200.195.xxx.xxx:47747
-- SIP/w2417-00000005 is busy
== Everyone is busy/congested at this time (1:1/0/0)
Please, can anybody show us some north?
Original comment by di...@fluxoti.com
on 26 Mar 2015 at 2:51
Same error here. ASterisk 1.8 vicidial
Original comment by OmarRodr...@gmail.com
on 14 May 2015 at 7:45
Hello,
I am getting same problem that is when I am making call from imsdroid to web
browser its showing the "failed to get local SDP offer". i am enable the RTCWeb
Breaker but problem not solve.
sip server kamailio
browser chrome Version 44.0.2403.155 (64-bit)
os ubuntu 14.04
sip client sipml5
softphone yureka (lollipop)
imsdroid version imsdroid-2.548.870.apk
Original comment by saurabhv...@gmail.com
on 19 Aug 2015 at 6:54
Original issue reported on code.google.com by
jaballah...@gmail.com
on 13 Nov 2012 at 1:51Attachments: