Issues
- 3
Inbound call is not working with Chrome 35
#179 opened by GoogleCodeExporter - 1
- 0
- 1
- 1
Regarding sipml api query
#174 opened by GoogleCodeExporter - 2
- 0
- 3
SIPML5 Error with multi-line support
#173 opened by GoogleCodeExporter - 1
Outgoing call from Doubango native clients to chrome using opus produce noise
#170 opened by GoogleCodeExporter - 2
Allow choosing the WebRTC implementation type (native, bowser or w4a)
#171 opened by GoogleCodeExporter - 1
Early audio are supported ?
#168 opened by GoogleCodeExporter - 1
- 0
Receive Bandwidth is too high
#165 opened by GoogleCodeExporter - 2
adds support for rfc5168
#166 opened by GoogleCodeExporter - 0
Call hangs on party hangup
#167 opened by GoogleCodeExporter - 2
- 1
SIPml5 declares a global MD5 object which may cause conflict with third party libraries
#164 opened by GoogleCodeExporter - 4
Crash IE 11
#161 opened by GoogleCodeExporter - 3
One Way video on VCS cisco
#162 opened by GoogleCodeExporter - 1
Crash IE 11
#160 opened by GoogleCodeExporter - 0
Video and audio problem on mobile device
#157 opened by GoogleCodeExporter - 1
setRemoteDescription Error
#158 opened by GoogleCodeExporter - 1
cache media stream don't works
#159 opened by GoogleCodeExporter - 4
Asterisk 11.7.0 + sipml5 + Chrome : one way RTP, no RTP from Chrome
#156 opened by GoogleCodeExporter - 2
Getting - SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.
#153 opened by GoogleCodeExporter - 0
forbidden and Not acceptable here calls
#154 opened by GoogleCodeExporter - 0
- 1
- 3
- 2
Firefox: removeStream not implemented yet
#150 opened by GoogleCodeExporter - 0
- 1
Patch for /trunk/error.htm
#147 opened by GoogleCodeExporter - 2
sipml5 live demo does not work when the ice turn is switched on with Firefox browser
#148 opened by GoogleCodeExporter - 3
- 1
- 2
sipML5 Hold/Resume/Transfer
#144 opened by GoogleCodeExporter - 2
sipml5 webrtc not able to hold a call or transfer it using Kamailio
#141 opened by GoogleCodeExporter - 1
Unable to register to FreeSwitch server & unable to call SIP client (XLite) respectively using SIPml5 client
#142 opened by GoogleCodeExporter - 0
crashes on providing dtmf during call
#143 opened by GoogleCodeExporter - 0
PSTN calls are forbidden
#139 opened by GoogleCodeExporter - 0
how to use sip option in sipml5
#140 opened by GoogleCodeExporter - 4
The delay from allow mic and start call to send invite delayed aoubt 10 secondes
#137 opened by GoogleCodeExporter - 0
Resume button is not working.... Please tell me the method which generates re-invite.
#138 opened by GoogleCodeExporter - 0
Cant get demo running
#133 opened by GoogleCodeExporter - 0
Display Name in From header not propagated to getRemoteFriendlyName()
#134 opened by GoogleCodeExporter - 0
- 0
central sip does not connect
#136 opened by GoogleCodeExporter - 23
Call established but no audio on both ends
#132 opened by GoogleCodeExporter - 1
the audio input in chrome
#131 opened by GoogleCodeExporter - 5