smallSquirrel / sipml5

Automatically exported from code.google.com/p/sipml5

smallSquirrel/sipml5 Issues

call disconnection in api version SIPML5 API version = 1.5.230
Updated 9 years ago
One way video, any browser, webrtc2sip and Asterisk to softphone
Updated 9 years ago2
Call.info() result is logged to console instead of given to user
Closed 9 years ago1
Temporary failure in name resolution
Updated 9 years ago
Problem in BYE send from SIPML5 - Webrtc2sip (Version 2.6.0)
Updated 9 years ago
Outbound call hangs up after 1 minute with FreeSwitch Session-Expire
Updated 9 years ago
Parameters send to Event Listener Function
Updated 9 years ago2
No early audio in outgoing sipml call to an extension.
Updated 9 years ago2
sipml5 unable to play remote audio
Updated 9 years ago2
Available on main package managers
Updated 9 years ago
Export to GitHub
Updated 9 years ago
Video stream doesnt add in SipML5
Updated 9 years ago
Firefox ICE resolution to incorrect 0.0.0.0 address
Closed 9 years ago2
Content length for SIP INFO packets is incorrect in latest release
Closed 9 years ago1
startBfcpShare not working
Updated 9 years ago
Called in wrong state: STATE_INPROGRESS
Updated 9 years ago7
dtmf('#') and dtmf('*') are not detected correctly
Updated 9 years ago
No Audio on Firefox 34+
Updated 9 years ago
Sipml5 not connecting
Closed 9 years ago1
SUBSCRIBE does not contain route headers
Updated 9 years ago
Double call "onGetUserMediaSuccess" and "createAnswer". Sipml5 can't answer on some calls and sends BYE.
Updated 9 years ago4
tsip_transac_layer.prototype.cancel_by_dialog incorrectly deletes transactions when there is more than one transaction
Closed 9 years ago2
Too large SIP INVITE
Updated 9 years ago
dtmf formatted incorrectly
Closed 9 years ago3
enable_media_stream_cache setting does not work
Updated 9 years ago2
SIP INFO DTMF not compatible with Asterisk
Closed 9 years ago1
webrtc4all is not detected
Updated 9 years ago1
Bad invite send by client on Chrome 39 (linux 64)
Closed 9 years ago1
Call between chrome to softphone call automatically dropped
Closed 9 years ago1
Compatibility problem with Node-Webkit?
Updated 9 years ago
Not working in safari 5.1.7
Closed 9 years ago1
Offer message containing c=IN IP4 0.0.0.0
Closed 9 years ago3
SipML5 + Asterisk : Remote host can't match request ACK
Closed 9 years ago4
Is 407 supported by sipml5
Closed 9 years ago2
WebRTC video call rejected asterisk11.130+sipml5
Closed 9 years ago1
which server can work with sipML5 api
Updated 9 years ago
Race condition in hold/resume
Updated 9 years ago1
sip call from android uses loud speaker instead of front speaker.
Updated 9 years ago2
SIPml5 behind NAT, Freeswitch on Amazon EC2, Calll Hang Up
Closed 9 years ago2
Asterisk on AWS; RTP reciving at client end but no voice
Updated 9 years ago1
SUBSCRIBE request fails \failing to find dialog
Updated 9 years ago7
No video on Firefox 30 :onIceGatheringCompleted but no local sdp request is pending
Updated 9 years ago
Settings ice_servers to [] does not disable ICE
Updated 9 years ago3
Incomming call without Video payload is refused
Updated 9 years ago
UPD flows bidirectionally but one way audio on webphone
Updated 9 years ago3
Unable to Login to OfficeSip Server using chrome v. 32
Updated 9 years ago1
Chrome m35(forces DTLS by default) <-> asterisk 11.9 [fix]
Updated 9 years ago6
Sipml5 SIP OPTIONS response SIP/2.0 405 Method Not Allowed
Updated 9 years ago
Transfer call functionality is not working as expected in sipml5
Updated 9 years ago1
WebRTC Not working with Google Chrome 35.
Closed 9 years ago9