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Arshimet
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sipml5
Automatically exported from code.google.com/p/sipml5
BSD 3-Clause "New" or "Revised" License
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SIPML5 Presence status changes are not reflect in desktop client
#77
GoogleCodeExporter
closed
9 years ago
2
One way audio and video with Google Chrome 25.0.1364.99
#76
GoogleCodeExporter
closed
9 years ago
13
Firefox/Chrome interop without webrtc2sip
#75
GoogleCodeExporter
opened
9 years ago
0
WebRTC API has been updated Options
#74
GoogleCodeExporter
opened
9 years ago
0
RTCPeerConnection API will update soon
#73
GoogleCodeExporter
opened
9 years ago
0
“wss” is Not a Valid Transport
#72
GoogleCodeExporter
opened
9 years ago
5
Chrome 24 does not play the remote audio on incoming calls
#71
GoogleCodeExporter
closed
9 years ago
2
Username in the SIP Request Line Missing
#70
GoogleCodeExporter
opened
9 years ago
1
Registration to Freeswitch ok, dialing not possible
#69
GoogleCodeExporter
opened
9 years ago
3
Unable to register as well as call an IMS UE
#68
GoogleCodeExporter
closed
9 years ago
4
Adds support for mute/unmute
#67
GoogleCodeExporter
opened
9 years ago
4
Add support for Firefox Nightly
#66
GoogleCodeExporter
closed
9 years ago
1
Add support for RFC 5939
#65
GoogleCodeExporter
closed
9 years ago
1
Failed to handle new challenge :: === PUBLISH Dialog terminated ===
#64
GoogleCodeExporter
opened
9 years ago
4
Add jsLint in the release process
#63
GoogleCodeExporter
opened
9 years ago
0
Send DTMF from sipml5 to ( webrtc2sip -> Asterisk 1.6/1.8 ) didnt work, nothing happened
#62
GoogleCodeExporter
closed
9 years ago
3
Something goes wrong when try to checkout Asterisk & sipml5 source code
#61
GoogleCodeExporter
closed
9 years ago
1
Make SIPml-api compliant for audio only
#60
GoogleCodeExporter
closed
9 years ago
2
Uncaught ReferenceError: tsip_header_get_name is not defined
#59
GoogleCodeExporter
closed
9 years ago
8
Use single js file for redistribution
#58
GoogleCodeExporter
closed
9 years ago
1
Unable to publish the presence status
#57
GoogleCodeExporter
closed
9 years ago
2
Failed to get local SDP offer
#56
GoogleCodeExporter
closed
9 years ago
21
Allow hacking the media profile using the 'expert' settings
#55
GoogleCodeExporter
opened
9 years ago
0
Error while video calling from IMSDroid(android) to SIPML5 on PC and vice vesa
#54
GoogleCodeExporter
closed
9 years ago
2
Inconsistent line ending style
#53
GoogleCodeExporter
closed
9 years ago
1
"Failed to create looper" during call
#52
GoogleCodeExporter
closed
9 years ago
1
webrtc4all mode is not detected in Opera 12.x
#51
GoogleCodeExporter
opened
9 years ago
0
Ericsson browser: "Failed to parse remote sdp"
#50
GoogleCodeExporter
opened
9 years ago
2
Add support for Bowser (Ericsson's WebRTC implementation)
#49
GoogleCodeExporter
opened
9 years ago
2
chrome to Asterisk call: SYNTAX_ERR: DOM Exception 12 upon 200 OK from Asterisk
#48
GoogleCodeExporter
closed
9 years ago
2
DomEX when a non ip-phone number call me
#47
GoogleCodeExporter
closed
9 years ago
3
Add support for GWT
#46
GoogleCodeExporter
closed
9 years ago
1
Failed to parse (remote) sdp message
#45
GoogleCodeExporter
closed
9 years ago
3
Call from chrome to softphone through asterisk cause only noise
#44
GoogleCodeExporter
closed
9 years ago
40
Hangup does not work if I dial
#43
GoogleCodeExporter
closed
9 years ago
3
SIPML5 + Asterisk - No audio after 30 seconds
#42
GoogleCodeExporter
closed
9 years ago
1
Bad quality of video when implement "sipML5 solution architecture (2)"
#41
GoogleCodeExporter
closed
9 years ago
5
Adds support for "webkitRTCPeerConnection" on chrome
#40
GoogleCodeExporter
closed
9 years ago
1
No audio with Freeswitch
#39
GoogleCodeExporter
closed
9 years ago
3
No Audio at all after updating to latest Asterisk Patch
#38
GoogleCodeExporter
opened
9 years ago
6
SYNTAX_ERR: DOM Exception 12 When Incoming call comes
#37
GoogleCodeExporter
closed
9 years ago
4
Inbound calls from Asterisk fail
#36
GoogleCodeExporter
closed
9 years ago
2
How to use video call from chrome to chrome and chrome to eyebeam or xlite5
#35
GoogleCodeExporter
closed
9 years ago
2
INVITE server transaction OK retransmissions don't stop after ACK
#34
GoogleCodeExporter
closed
9 years ago
3
load specific revision?
#33
GoogleCodeExporter
closed
9 years ago
1
Make it more like library
#32
GoogleCodeExporter
closed
9 years ago
7
www.sipML5.org/call.htm doesn't work!
#31
GoogleCodeExporter
closed
9 years ago
1
onclose event not raised when websocket is closed
#30
GoogleCodeExporter
opened
9 years ago
0
tsip_message_parser do not parse message with non english content
#29
GoogleCodeExporter
closed
9 years ago
3
Installation of source
#28
GoogleCodeExporter
closed
9 years ago
4
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