RestComm webrtcomm issues

RestComm / webrtcomm

WebRTCComm is a simple high level JavaScript WebRTC framework for Web Developers to add Real Time Communications and IM Capabilities to any website.

http://www.restcomm.com/

GNU General Public License v3.0

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issues

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Fixed FS-142: Avoid sporadic 40-second delay when making calls with Olympus

#107 atsakiridis opened 6 years ago
0
Incorrect handling of headers with multiple occurrences

#106 ammendonca closed 6 years ago
0
Support Safari 11 which supports webrtc/h264

#105 atsakiridis opened 6 years ago
0
Webrtcomm shouldn't ACKs a 603 Declined response to its INVITE

#104 atsakiridis opened 7 years ago
0
iceServers not properly populated when coming from a configuration url (like Xirsys)

#103 atsakiridis closed 7 years ago
0
addition to #43 - demo NodeJS webapp

#102 akdeniso opened 7 years ago
2
Consider hosting sample/demo applications at GitHub

#101 atsakiridis opened 7 years ago
0
Fix 'Call already ongoing'

#100 atsakiridis opened 7 years ago
1
Fix normalization of webrtc media stats that got broken

#99 atsakiridis opened 7 years ago
0
Always log webrtc getStats when a call is hung up

#98 atsakiridis closed 7 years ago
0
Introduce number validation in SDK

#97 atsakiridis opened 7 years ago
0
486 Busy here shouldn't be shown in the logs are an error

#96 atsakiridis opened 7 years ago
0
Merge FSM for SIP_INVITING_STATE and SIP_INVITING_407_STATE

#95 ammendonca closed 7 years ago
1
Exception when callee hangs up outbound call

#94 atsakiridis opened 7 years ago
0
Update documentation on ICE servers

#93 atsakiridis closed 7 years ago
0
Introduce remote logging for cloud installations

#92 atsakiridis opened 7 years ago
0
Make version numbering manually configurable

#91 atsakiridis opened 7 years ago
1
When making a call, if we get 401, we should handle it same way as 407 and provide credentials

#90 ocarriles opened 7 years ago
12
Fixed #43: Introduce sample screen sharing webapp

#89 akdeniso opened 7 years ago
0
Final touches on SDK for Release Candidate 1

#88 atsakiridis opened 7 years ago
0
Introduce Integration Testing

#87 atsakiridis opened 7 years ago
0
Introduce Unit Testing

#86 atsakiridis opened 7 years ago
0
Trying to add epic label

#85 atsakiridis closed 7 years ago
0
Increase registration refresh + expiration interval

#84 atsakiridis closed 7 years ago
0
Check if current keep-alive mechanism employed by Restcomm Connect is enough to keep NAT holes open

#83 atsakiridis opened 7 years ago
0
Consider improving received ACK handling after 200 OK in UAS

#82 atsakiridis opened 7 years ago
0
Integrate SDK with Restcomm Statistics

#81 atsakiridis opened 7 years ago
0
Avoid reseting PrivateJainSipClientConnector if a Register fails

#80 atsakiridis closed 7 years ago
0
Reject an incoming call if there is an ongoing Call

#79 atsakiridis closed 7 years ago
0
Test RTCDTMFSender for DTMF transmission for Firefox when it becomes available

#78 atsakiridis opened 7 years ago
0
Implement media handover between Wifi and Cellular Data for a live call

#77 atsakiridis opened 7 years ago
0
Enhance logging so that log levels are clear even when investigating logs outside of a browser

#76 atsakiridis closed 7 years ago
0
There are a lot of cases where an error happens, but the log emitted is type debug

#75 atsakiridis closed 7 years ago
0
Fix build.sh to avoid removing all the build dir

#74 atsakiridis closed 7 years ago
1
We shouldn't need an additional variable to hold Peer Connection state

#73 atsakiridis opened 7 years ago
0
We shouldn't get a warning when the call is hung up for iceconnectionstatechange

#72 atsakiridis closed 7 years ago
0
Add better Peer Connection logging

#71 atsakiridis closed 7 years ago
0
Refactor Peer Connection callbacks

#70 atsakiridis closed 7 years ago
0
In webrtcomm sample use symbolic links for webrtcomm and jain-sip js libs

#69 atsakiridis closed 7 years ago
0
Add guard so that if a call is already ongoing we don't allow another one to be made

#68 atsakiridis closed 7 years ago
0
Default to SIP INFO for DTMF digit transmission, instead of RTP

#67 atsakiridis closed 7 years ago
1
Refactor onWebRTCommCallOpenedEvent

#66 atsakiridis closed 7 years ago
2
Enhance logs so that browser + version is shown

#65 atsakiridis opened 7 years ago
0
webrtcomm is displaying password in cleartext in logs

#64 deruelle closed 7 years ago
0
Fix issue with bad PeerConnection state

#63 atsakiridis closed 7 years ago
2
Consider automatically unregistering if the user closes the Olympus browser tab

#62 atsakiridis closed 7 years ago
1
Consider a mechanism to reconnect in the event that WS connectivity is lost

#61 atsakiridis opened 7 years ago
0
Consider providing a way to adjust webrtc bandwidth in the SDK

#60 atsakiridis opened 8 years ago
0
Webrtcomm can't handle 100 Trying after sending MESSAGE

#59 atsakiridis closed 7 years ago
1
Add reason header to BYE

#58 atsakiridis opened 8 years ago
0