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DoubangoTelecom
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sipml5
The world's first HTML5 SIP client (WebRTC)
BSD 3-Clause "New" or "Revised" License
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Firefox ICE resolution to incorrect 0.0.0.0 address
#217
GoogleCodeExporter
closed
9 years ago
2
Content length for SIP INFO packets is incorrect in latest release
#216
GoogleCodeExporter
closed
9 years ago
1
startBfcpShare not working
#215
GoogleCodeExporter
opened
9 years ago
0
Called in wrong state: STATE_INPROGRESS
#214
GoogleCodeExporter
opened
9 years ago
10
dtmf('#') and dtmf('*') are not detected correctly
#213
GoogleCodeExporter
opened
9 years ago
0
No Audio on Firefox 34+
#212
GoogleCodeExporter
opened
9 years ago
0
Sipml5 not connecting
#211
GoogleCodeExporter
closed
9 years ago
1
SUBSCRIBE does not contain route headers
#210
GoogleCodeExporter
opened
9 years ago
0
Double call "onGetUserMediaSuccess" and "createAnswer". Sipml5 can't answer on some calls and sends BYE.
#209
GoogleCodeExporter
opened
9 years ago
5
tsip_transac_layer.prototype.cancel_by_dialog incorrectly deletes transactions when there is more than one transaction
#208
GoogleCodeExporter
closed
9 years ago
2
Too large SIP INVITE
#207
GoogleCodeExporter
opened
9 years ago
0
dtmf formatted incorrectly
#206
GoogleCodeExporter
closed
9 years ago
3
enable_media_stream_cache setting does not work
#205
GoogleCodeExporter
opened
9 years ago
2
SIP INFO DTMF not compatible with Asterisk
#204
GoogleCodeExporter
closed
9 years ago
1
webrtc4all is not detected
#203
GoogleCodeExporter
closed
8 years ago
2
Bad invite send by client on Chrome 39 (linux 64)
#202
GoogleCodeExporter
closed
9 years ago
1
Call between chrome to softphone call automatically dropped
#201
GoogleCodeExporter
closed
9 years ago
1
Compatibility problem with Node-Webkit?
#200
GoogleCodeExporter
opened
9 years ago
0
Not working in safari 5.1.7
#199
GoogleCodeExporter
closed
9 years ago
1
Offer message containing c=IN IP4 0.0.0.0
#198
GoogleCodeExporter
closed
9 years ago
3
SipML5 + Asterisk : Remote host can't match request ACK
#197
GoogleCodeExporter
closed
9 years ago
4
Is 407 supported by sipml5
#196
GoogleCodeExporter
closed
9 years ago
2
WebRTC video call rejected asterisk11.130+sipml5
#195
GoogleCodeExporter
closed
9 years ago
1
which server can work with sipML5 api
#194
GoogleCodeExporter
opened
9 years ago
0
Race condition in hold/resume
#193
GoogleCodeExporter
opened
9 years ago
1
sip call from android uses loud speaker instead of front speaker.
#192
GoogleCodeExporter
opened
9 years ago
2
SIPml5 behind NAT, Freeswitch on Amazon EC2, Calll Hang Up
#191
GoogleCodeExporter
closed
9 years ago
2
Asterisk on AWS; RTP reciving at client end but no voice
#190
GoogleCodeExporter
opened
9 years ago
1
SUBSCRIBE request fails \failing to find dialog
#189
GoogleCodeExporter
opened
9 years ago
7
No video on Firefox 30 :onIceGatheringCompleted but no local sdp request is pending
#188
GoogleCodeExporter
opened
9 years ago
0
Settings ice_servers to [] does not disable ICE
#187
GoogleCodeExporter
opened
9 years ago
3
Incomming call without Video payload is refused
#186
GoogleCodeExporter
opened
9 years ago
0
UPD flows bidirectionally but one way audio on webphone
#185
GoogleCodeExporter
opened
9 years ago
3
Unable to Login to OfficeSip Server using chrome v. 32
#184
GoogleCodeExporter
opened
9 years ago
1
Chrome m35(forces DTLS by default) <-> asterisk 11.9 [fix]
#183
GoogleCodeExporter
opened
9 years ago
6
Sipml5 SIP OPTIONS response SIP/2.0 405 Method Not Allowed
#182
GoogleCodeExporter
opened
9 years ago
0
Transfer call functionality is not working as expected in sipml5
#181
GoogleCodeExporter
opened
9 years ago
1
WebRTC Not working with Google Chrome 35.
#180
GoogleCodeExporter
closed
9 years ago
9
Inbound call is not working with Chrome 35
#179
GoogleCodeExporter
closed
9 years ago
3
'stopping' event never signalled, 'stopped' signalled twice
#178
GoogleCodeExporter
opened
9 years ago
0
IE 11 bug with webrtc4all: fakelooper is undefined
#177
GoogleCodeExporter
opened
9 years ago
1
SUBSCRIBE session restarts expire timer after each NOTIFY
#176
GoogleCodeExporter
opened
9 years ago
1
sdp error: Failed to parse audio codecs correctly
#175
GoogleCodeExporter
closed
9 years ago
2
Regarding sipml api query
#174
GoogleCodeExporter
closed
9 years ago
1
SIPML5 Error with multi-line support
#173
GoogleCodeExporter
closed
9 years ago
3
cannot igore the reinvite request from freeswitch
#172
GoogleCodeExporter
opened
9 years ago
0
Allow choosing the WebRTC implementation type (native, bowser or w4a)
#171
GoogleCodeExporter
closed
9 years ago
2
Outgoing call from Doubango native clients to chrome using opus produce noise
#170
GoogleCodeExporter
opened
9 years ago
1
Use "UDP/TLS/RTP/SAVPF" when "a=crypto" attributes are missing
#169
GoogleCodeExporter
closed
9 years ago
1
Early audio are supported ?
#168
GoogleCodeExporter
closed
9 years ago
1
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