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sipml5
Automatically exported from code.google.com/p/sipml5
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Chrome m35(forces DTLS by default) <-> asterisk 11.9 [fix]
#183
GoogleCodeExporter
opened
8 years ago
6
Sipml5 SIP OPTIONS response SIP/2.0 405 Method Not Allowed
#182
GoogleCodeExporter
opened
8 years ago
0
Transfer call functionality is not working as expected in sipml5
#181
GoogleCodeExporter
opened
8 years ago
1
WebRTC Not working with Google Chrome 35.
#180
GoogleCodeExporter
closed
8 years ago
9
Inbound call is not working with Chrome 35
#179
GoogleCodeExporter
closed
8 years ago
3
'stopping' event never signalled, 'stopped' signalled twice
#178
GoogleCodeExporter
opened
8 years ago
0
IE 11 bug with webrtc4all: fakelooper is undefined
#177
GoogleCodeExporter
opened
8 years ago
1
SUBSCRIBE session restarts expire timer after each NOTIFY
#176
GoogleCodeExporter
opened
8 years ago
1
sdp error: Failed to parse audio codecs correctly
#175
GoogleCodeExporter
closed
8 years ago
2
Regarding sipml api query
#174
GoogleCodeExporter
closed
8 years ago
1
SIPML5 Error with multi-line support
#173
GoogleCodeExporter
closed
8 years ago
3
cannot igore the reinvite request from freeswitch
#172
GoogleCodeExporter
opened
8 years ago
0
Allow choosing the WebRTC implementation type (native, bowser or w4a)
#171
GoogleCodeExporter
closed
8 years ago
2
Outgoing call from Doubango native clients to chrome using opus produce noise
#170
GoogleCodeExporter
opened
8 years ago
1
Use "UDP/TLS/RTP/SAVPF" when "a=crypto" attributes are missing
#169
GoogleCodeExporter
closed
8 years ago
1
Early audio are supported ?
#168
GoogleCodeExporter
closed
8 years ago
1
Call hangs on party hangup
#167
GoogleCodeExporter
opened
8 years ago
0
adds support for rfc5168
#166
GoogleCodeExporter
closed
8 years ago
2
Receive Bandwidth is too high
#165
GoogleCodeExporter
opened
8 years ago
0
SIPml5 declares a global MD5 object which may cause conflict with third party libraries
#164
GoogleCodeExporter
opened
8 years ago
1
Caller ID Name Not Present on Incoming Call to sipML5
#163
GoogleCodeExporter
opened
8 years ago
2
One Way video on VCS cisco
#162
GoogleCodeExporter
closed
8 years ago
3
Crash IE 11
#161
GoogleCodeExporter
closed
8 years ago
4
Crash IE 11
#160
GoogleCodeExporter
closed
8 years ago
1
cache media stream don't works
#159
GoogleCodeExporter
opened
8 years ago
1
setRemoteDescription Error
#158
GoogleCodeExporter
closed
8 years ago
1
Video and audio problem on mobile device
#157
GoogleCodeExporter
opened
8 years ago
0
Asterisk 11.7.0 + sipml5 + Chrome : one way RTP, no RTP from Chrome
#156
GoogleCodeExporter
opened
8 years ago
4
Sipml5 + office sip server -unable to login
#155
GoogleCodeExporter
opened
8 years ago
0
forbidden and Not acceptable here calls
#154
GoogleCodeExporter
opened
8 years ago
0
Getting - SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.
#153
GoogleCodeExporter
closed
8 years ago
2
enable_media_stream_cache option doesn't work
#152
GoogleCodeExporter
closed
8 years ago
3
Disconnected: Failed to connet to the server
#151
GoogleCodeExporter
opened
8 years ago
1
Firefox: removeStream not implemented yet
#150
GoogleCodeExporter
closed
8 years ago
2
WS connection continues after losing internet connection
#149
GoogleCodeExporter
opened
8 years ago
0
sipml5 live demo does not work when the ice turn is switched on with Firefox browser
#148
GoogleCodeExporter
opened
8 years ago
2
Patch for /trunk/error.htm
#147
GoogleCodeExporter
closed
8 years ago
1
can't answering incomming call after forcing connection for user
#146
GoogleCodeExporter
opened
8 years ago
1
SipML5 Expert mode: Max bandwidth (kbps) doesn't work!
#145
GoogleCodeExporter
closed
8 years ago
3
sipML5 Hold/Resume/Transfer
#144
GoogleCodeExporter
closed
8 years ago
2
crashes on providing dtmf during call
#143
GoogleCodeExporter
opened
8 years ago
0
Unable to register to FreeSwitch server & unable to call SIP client (XLite) respectively using SIPml5 client
#142
GoogleCodeExporter
opened
8 years ago
1
sipml5 webrtc not able to hold a call or transfer it using Kamailio
#141
GoogleCodeExporter
opened
8 years ago
2
how to use sip option in sipml5
#140
GoogleCodeExporter
opened
8 years ago
0
PSTN calls are forbidden
#139
GoogleCodeExporter
opened
8 years ago
0
Resume button is not working.... Please tell me the method which generates re-invite.
#138
GoogleCodeExporter
opened
8 years ago
0
The delay from allow mic and start call to send invite delayed aoubt 10 secondes
#137
GoogleCodeExporter
opened
8 years ago
4
central sip does not connect
#136
GoogleCodeExporter
opened
8 years ago
0
Retransmission of ACK - Callee keeps receiving ACK .
#135
GoogleCodeExporter
opened
8 years ago
0
Display Name in From header not propagated to getRemoteFriendlyName()
#134
GoogleCodeExporter
opened
8 years ago
0
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