jefferson2392 sipml5 issues

jefferson2392 / 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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