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hemaxox / sipml5

Automatically exported from code.google.com/p/sipml5
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Failed to get local SDP offer #56

Closed GoogleCodeExporter closed 8 years ago

GoogleCodeExporter commented 8 years ago 
hello,
i'm using ubuntu / chrome 23

i'm using the live demo. 

When i receive a call from a phone i get an arror : Failed to get local SDP 
offer.
i'm attaching the error from the console.

thank you for any help.

Original issue reported on code.google.com by jaballah...@gmail.com on 13 Nov 2012 at 1:51

Attachments:

GoogleCodeExporter commented 8 years ago 
How can i fix this error ?

Original comment by jaballah...@gmail.com on 27 Nov 2012 at 8:00

GoogleCodeExporter commented 8 years ago 
[deleted comment]
GoogleCodeExporter commented 8 years ago 
Anything new int this issue?

Original comment by Jakub.Tr...@gmail.com on 2 Dec 2012 at 10:27

GoogleCodeExporter commented 8 years ago 
No, i still have this issue ...

Original comment by jaballah...@gmail.com on 3 Dec 2012 at 7:29

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
Yes, have this issue on Mac too

Original comment by Jakub.Tr...@gmail.com on 3 Dec 2012 at 8:44

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
@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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
attaching javascript console log.

Original comment by akxte...@gmail.com on 28 Dec 2012 at 6:27

Attachments:

GoogleCodeExporter commented 8 years ago 
Windows 7 64 Chrome, I see this bug too (

Original comment by 4587...@gmail.com on 31 Dec 2012 at 8:05

GoogleCodeExporter commented 8 years ago 
@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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
@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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
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

GoogleCodeExporter commented 8 years ago 
Same error here. ASterisk 1.8 vicidial

Original comment by OmarRodr...@gmail.com on 14 May 2015 at 7:45

GoogleCodeExporter commented 8 years ago 
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