github-petr-novak sipml5 issues

github-petr-novak / sipml5

Automatically exported from code.google.com/p/sipml5

BSD 3-Clause "New" or "Revised" License

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issues

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Could not set SRTP policies

#83 GoogleCodeExporter opened 8 years ago
5
BYE not received

#82 GoogleCodeExporter opened 8 years ago
3
sipml5 does not work in firefox,chrome

#81 GoogleCodeExporter closed 8 years ago
6
Sipml5 - OpenIMSCore: Bad Request - Not following indicated Service-Routes

#80 GoogleCodeExporter closed 8 years ago
4
RTCWeb browsers compatibilities

#79 GoogleCodeExporter closed 8 years ago
2
sipml5 SetRemoteDescription failed

#78 GoogleCodeExporter closed 8 years ago
6
SIPML5 Presence status changes are not reflect in desktop client

#77 GoogleCodeExporter closed 8 years ago
2
One way audio and video with Google Chrome 25.0.1364.99

#76 GoogleCodeExporter closed 8 years ago
13
Firefox/Chrome interop without webrtc2sip

#75 GoogleCodeExporter opened 8 years ago
0
WebRTC API has been updated Options

#74 GoogleCodeExporter opened 8 years ago
0
RTCPeerConnection API will update soon

#73 GoogleCodeExporter opened 8 years ago
0
“wss” is Not a Valid Transport

#72 GoogleCodeExporter opened 8 years ago
5
Chrome 24 does not play the remote audio on incoming calls

#71 GoogleCodeExporter closed 8 years ago
2
Username in the SIP Request Line Missing

#70 GoogleCodeExporter opened 8 years ago
1
Registration to Freeswitch ok, dialing not possible

#69 GoogleCodeExporter opened 8 years ago
4
Unable to register as well as call an IMS UE

#68 GoogleCodeExporter closed 8 years ago
4
Adds support for mute/unmute

#67 GoogleCodeExporter opened 8 years ago
4
Add support for Firefox Nightly

#66 GoogleCodeExporter closed 8 years ago
1
Add support for RFC 5939

#65 GoogleCodeExporter closed 8 years ago
1
Failed to handle new challenge :: === PUBLISH Dialog terminated ===

#64 GoogleCodeExporter opened 8 years ago
4
Add jsLint in the release process

#63 GoogleCodeExporter opened 8 years ago
0
Send DTMF from sipml5 to ( webrtc2sip -> Asterisk 1.6/1.8 ) didnt work, nothing happened

#62 GoogleCodeExporter closed 8 years ago
3
Something goes wrong when try to checkout Asterisk & sipml5 source code

#61 GoogleCodeExporter closed 8 years ago
1
Make SIPml-api compliant for audio only

#60 GoogleCodeExporter closed 8 years ago
2
Uncaught ReferenceError: tsip_header_get_name is not defined

#59 GoogleCodeExporter closed 8 years ago
8
Use single js file for redistribution

#58 GoogleCodeExporter closed 8 years ago
1
Unable to publish the presence status

#57 GoogleCodeExporter closed 8 years ago
2
Failed to get local SDP offer

#56 GoogleCodeExporter closed 8 years ago
22
Allow hacking the media profile using the 'expert' settings

#55 GoogleCodeExporter opened 8 years ago
0
Error while video calling from IMSDroid(android) to SIPML5 on PC and vice vesa

#54 GoogleCodeExporter closed 8 years ago
2
Inconsistent line ending style

#53 GoogleCodeExporter closed 8 years ago
1
"Failed to create looper" during call

#52 GoogleCodeExporter closed 8 years ago
1
webrtc4all mode is not detected in Opera 12.x

#51 GoogleCodeExporter opened 8 years ago
0
Ericsson browser: "Failed to parse remote sdp"

#50 GoogleCodeExporter opened 8 years ago
2
Add support for Bowser (Ericsson's WebRTC implementation)

#49 GoogleCodeExporter opened 8 years ago
2
chrome to Asterisk call: SYNTAX_ERR: DOM Exception 12 upon 200 OK from Asterisk

#48 GoogleCodeExporter closed 8 years ago
2
DomEX when a non ip-phone number call me

#47 GoogleCodeExporter closed 8 years ago
3
Add support for GWT

#46 GoogleCodeExporter closed 8 years ago
1
Failed to parse (remote) sdp message

#45 GoogleCodeExporter closed 8 years ago
3
Call from chrome to softphone through asterisk cause only noise

#44 GoogleCodeExporter closed 8 years ago
40
Hangup does not work if I dial

#43 GoogleCodeExporter closed 8 years ago
3
SIPML5 + Asterisk - No audio after 30 seconds

#42 GoogleCodeExporter closed 8 years ago
1
Bad quality of video when implement "sipML5 solution architecture (2)"

#41 GoogleCodeExporter closed 8 years ago
5
Adds support for "webkitRTCPeerConnection" on chrome

#40 GoogleCodeExporter closed 8 years ago
1
No audio with Freeswitch

#39 GoogleCodeExporter closed 8 years ago
3
No Audio at all after updating to latest Asterisk Patch

#38 GoogleCodeExporter opened 8 years ago
6
SYNTAX_ERR: DOM Exception 12 When Incoming call comes

#37 GoogleCodeExporter closed 8 years ago
4
Inbound calls from Asterisk fail

#36 GoogleCodeExporter closed 8 years ago
2
How to use video call from chrome to chrome and chrome to eyebeam or xlite5

#35 GoogleCodeExporter closed 8 years ago
2
INVITE server transaction OK retransmissions don't stop after ACK

#34 GoogleCodeExporter closed 8 years ago
3

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