patryksim Sipml5 issues

patryksim / Sipml5

Basado en la librería SIPML5, combinando las capacidades de WebRTC y el protocolo SIP para ofrecer comunicaciones de voz y video directamente desde un navegador web compatible.

BSD 3-Clause "New" or "Revised" License

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issues

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Call.info() result is logged to console instead of given to user

#227 GoogleCodeExporter closed 9 years ago
1
Temporary failure in name resolution

#226 GoogleCodeExporter opened 9 years ago
0
Problem in BYE send from SIPML5 - Webrtc2sip (Version 2.6.0)

#225 GoogleCodeExporter opened 9 years ago
0
Outbound call hangs up after 1 minute with FreeSwitch Session-Expire

#224 GoogleCodeExporter opened 9 years ago
0
Parameters send to Event Listener Function

#223 GoogleCodeExporter opened 9 years ago
2
No early audio in outgoing sipml call to an extension.

#222 GoogleCodeExporter opened 9 years ago
2
sipml5 unable to play remote audio

#221 GoogleCodeExporter opened 9 years ago
2
Available on main package managers

#220 GoogleCodeExporter opened 9 years ago
0
Export to GitHub

#219 GoogleCodeExporter opened 9 years ago
0
Video stream doesnt add in SipML5

#218 GoogleCodeExporter opened 9 years ago
0
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
7
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
4
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 opened 9 years ago
1
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
0
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