Ecosyste.ms: Issues
An open API service for providing issue and pull request metadata for open source projects.
GitHub / RestComm/webrtcomm issues and pull requests
#107 - Fixed FS-142: Avoid sporadic 40-second delay when making calls with Olympus
Pull Request -
State: open - Opened by atsakiridis over 6 years ago
#106 - Incorrect handling of headers with multiple occurrences
Issue -
State: closed - Opened by ammendonca almost 7 years ago
Labels: bug, important, unplanned
#105 - Support Safari 11 which supports webrtc/h264
Issue -
State: open - Opened by atsakiridis over 7 years ago
#104 - Webrtcomm shouldn't ACKs a 603 Declined response to its INVITE
Issue -
State: open - Opened by atsakiridis over 7 years ago
#103 - iceServers not properly populated when coming from a configuration url (like Xirsys)
Issue -
State: closed - Opened by atsakiridis over 7 years ago
Labels: unplanned
#102 - addition to #43 - demo NodeJS webapp
Pull Request -
State: open - Opened by akdeniso over 7 years ago
- 2 comments
#101 - Consider hosting sample/demo applications at GitHub
Issue -
State: open - Opened by atsakiridis over 7 years ago
#100 - Fix 'Call already ongoing'
Issue -
State: open - Opened by atsakiridis almost 8 years ago
- 1 comment
Labels: important
#99 - Fix normalization of webrtc media stats that got broken
Issue -
State: open - Opened by atsakiridis almost 8 years ago
#98 - Always log webrtc getStats when a call is hung up
Issue -
State: closed - Opened by atsakiridis almost 8 years ago
Labels: unplanned
#97 - Introduce number validation in SDK
Issue -
State: open - Opened by atsakiridis almost 8 years ago
#96 - 486 Busy here shouldn't be shown in the logs are an error
Issue -
State: open - Opened by atsakiridis almost 8 years ago
#95 - Merge FSM for SIP_INVITING_STATE and SIP_INVITING_407_STATE
Pull Request -
State: closed - Opened by ammendonca almost 8 years ago
- 1 comment
#94 - Exception when callee hangs up outbound call
Issue -
State: open - Opened by atsakiridis almost 8 years ago
Labels: bug
#93 - Update documentation on ICE servers
Issue -
State: closed - Opened by atsakiridis almost 8 years ago
#92 - Introduce remote logging for cloud installations
Issue -
State: open - Opened by atsakiridis almost 8 years ago
#91 - Make version numbering manually configurable
Issue -
State: open - Opened by atsakiridis almost 8 years ago
- 1 comment
#90 - When making a call, if we get 401, we should handle it same way as 407 and provide credentials
Issue -
State: open - Opened by ocarriles almost 8 years ago
- 12 comments
Labels: help wanted
#89 - Fixed #43: Introduce sample screen sharing webapp
Pull Request -
State: open - Opened by akdeniso almost 8 years ago
#88 - Final touches on SDK for Release Candidate 1
Issue -
State: open - Opened by atsakiridis almost 8 years ago
Labels: Epic
#87 - Introduce Integration Testing
Issue -
State: open - Opened by atsakiridis almost 8 years ago
Labels: Epic
#86 - Introduce Unit Testing
Issue -
State: open - Opened by atsakiridis almost 8 years ago
Labels: Epic
#85 - Trying to add epic label
Issue -
State: closed - Opened by atsakiridis almost 8 years ago
Labels: Epic
#84 - Increase registration refresh + expiration interval
Issue -
State: closed - Opened by atsakiridis about 8 years ago
#83 - Check if current keep-alive mechanism employed by Restcomm Connect is enough to keep NAT holes open
Issue -
State: open - Opened by atsakiridis about 8 years ago
#82 - Consider improving received ACK handling after 200 OK in UAS
Issue -
State: open - Opened by atsakiridis about 8 years ago
Labels: tech debt
#81 - Integrate SDK with Restcomm Statistics
Issue -
State: open - Opened by atsakiridis about 8 years ago
#80 - Avoid reseting PrivateJainSipClientConnector if a Register fails
Issue -
State: closed - Opened by atsakiridis about 8 years ago
Labels: important
#79 - Reject an incoming call if there is an ongoing Call
Issue -
State: closed - Opened by atsakiridis about 8 years ago
#78 - Test RTCDTMFSender for DTMF transmission for Firefox when it becomes available
Issue -
State: open - Opened by atsakiridis about 8 years ago
Labels: testing
#77 - Implement media handover between Wifi and Cellular Data for a live call
Issue -
State: open - Opened by atsakiridis about 8 years ago
#76 - Enhance logging so that log levels are clear even when investigating logs outside of a browser
Issue -
State: closed - Opened by atsakiridis about 8 years ago
Labels: important
#75 - There are a lot of cases where an error happens, but the log emitted is type debug
Issue -
State: closed - Opened by atsakiridis about 8 years ago
#74 - Fix build.sh to avoid removing all the build dir
Issue -
State: closed - Opened by atsakiridis about 8 years ago
- 1 comment
#73 - We shouldn't need an additional variable to hold Peer Connection state
Issue -
State: open - Opened by atsakiridis about 8 years ago
Labels: help wanted
#72 - We shouldn't get a warning when the call is hung up for iceconnectionstatechange
Issue -
State: closed - Opened by atsakiridis about 8 years ago
#71 - Add better Peer Connection logging
Issue -
State: closed - Opened by atsakiridis about 8 years ago
Labels: important
#70 - Refactor Peer Connection callbacks
Issue -
State: closed - Opened by atsakiridis about 8 years ago
Labels: important
#69 - In webrtcomm sample use symbolic links for webrtcomm and jain-sip js libs
Issue -
State: closed - Opened by atsakiridis about 8 years ago
#68 - Add guard so that if a call is already ongoing we don't allow another one to be made
Issue -
State: closed - Opened by atsakiridis about 8 years ago
Labels: important
#67 - Default to SIP INFO for DTMF digit transmission, instead of RTP
Issue -
State: closed - Opened by atsakiridis about 8 years ago
- 1 comment
#66 - Refactor onWebRTCommCallOpenedEvent
Issue -
State: closed - Opened by atsakiridis about 8 years ago
- 2 comments
Labels: important
#65 - Enhance logs so that browser + version is shown
Issue -
State: open - Opened by atsakiridis about 8 years ago
Labels: important
#64 - webrtcomm is displaying password in cleartext in logs
Issue -
State: closed - Opened by deruelle over 8 years ago
Labels: bug
#63 - Fix issue with bad PeerConnection state
Issue -
State: closed - Opened by atsakiridis over 8 years ago
- 2 comments
Labels: important
#62 - Consider automatically unregistering if the user closes the Olympus browser tab
Issue -
State: closed - Opened by atsakiridis over 8 years ago
- 1 comment
#61 - Consider a mechanism to reconnect in the event that WS connectivity is lost
Issue -
State: open - Opened by atsakiridis over 8 years ago
Labels: important
#60 - Consider providing a way to adjust webrtc bandwidth in the SDK
Issue -
State: open - Opened by atsakiridis over 8 years ago
#59 - Webrtcomm can't handle 100 Trying after sending MESSAGE
Issue -
State: closed - Opened by atsakiridis over 8 years ago
- 1 comment
Labels: help wanted
#58 - Add reason header to BYE
Issue -
State: open - Opened by atsakiridis over 8 years ago
#57 - Implement upgrade of media from audio -> audio/video
Issue -
State: open - Opened by atsakiridis over 8 years ago
#56 - Seems that Chrome is much slower than Firefox on webrtc call setup
Issue -
State: closed - Opened by atsakiridis over 8 years ago
- 1 comment
#55 - Add timestamp in logs to ease troubleshooting
Issue -
State: closed - Opened by atsakiridis over 8 years ago
- 1 comment
#54 - Registration refresh period and expiry are very far apart
Issue -
State: closed - Opened by atsakiridis over 8 years ago
#53 - Expose QoS stats in Restcomm-Connect
Issue -
State: open - Opened by atsakiridis over 8 years ago
#52 - Implement 3PCC scenario
Issue -
State: closed - Opened by atsakiridis over 8 years ago
- 1 comment
#51 - Add custom SIP headers for outgoing calls
Issue -
State: open - Opened by atsakiridis over 8 years ago
#50 - Introduce CI/CD in webrtcomm
Issue -
State: open - Opened by atsakiridis almost 9 years ago
- 3 comments
Labels: Epic
#49 - Document custom SIP headers for incoming calls
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
Labels: in progress
#48 - Add custom headers support for incoming calls
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
Labels: in progress
#47 - When a registration times out we stop trying in the future
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
- 1 comment
#46 - In logs also print out an identifier to correlate logs
Issue -
State: open - Opened by atsakiridis almost 9 years ago
#45 - Video Conferencing
Issue -
State: open - Opened by atsakiridis almost 9 years ago
#44 - Location support
Issue -
State: open - Opened by atsakiridis almost 9 years ago
#43 - Screen Sharing
Issue -
State: open - Opened by atsakiridis almost 9 years ago
- 3 comments
#42 - Server Based Group Chat
Issue -
State: open - Opened by atsakiridis almost 9 years ago
#41 - Recording
Issue -
State: open - Opened by atsakiridis almost 9 years ago
#40 - Seems IceTransports constraint breaks Chrome Beta build
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
- 1 comment
#39 - Support DTMF for Firefox
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
Labels: in progress
#38 - Document the getStats() API in reference documentation
Issue -
State: open - Opened by atsakiridis almost 9 years ago
Labels: documenting
#37 - Inconsistency between Chrome and Firefox in P2P call
Issue -
State: open - Opened by atsakiridis almost 9 years ago
- 1 comment
#36 - Add kbps metric for getStats for Chrome as well
Issue -
State: open - Opened by atsakiridis almost 9 years ago
- 1 comment
#35 - Implement on-demand getStats()
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
#34 - Indent SDK cause currently some sections are difficult to follow
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
#33 - Integrate webrtcomm with adapter.js
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
- 1 comment
#32 - add getStats() functionality in webrtcomm
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
#31 - Add new event type for issues that occur while on call
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
#30 - Fix build.sh for webrtcomm and jain-sip-js so that on non-debug .js libs only contain error logs
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
#29 - Replace node js http server code with python
Issue -
State: closed - Opened by atsakiridis almost 9 years ago
- 1 comment
#28 - Use version numbering in Webrtcomm library
Issue -
State: open - Opened by atsakiridis almost 9 years ago
- 2 comments
Labels: Epic
#27 - Consider implementing media level tracking
Issue -
State: open - Opened by atsakiridis about 9 years ago
#26 - Add ignore functionality in PrivateJainSipCallConnector
Issue -
State: closed - Opened by atsakiridis about 9 years ago
#25 - Notify WebRTCommCall when an incoming call is canceled
Issue -
State: closed - Opened by atsakiridis about 9 years ago
#24 - sipOutboundProxy and sipDomain should not be needed when in registrar-less mode
Issue -
State: open - Opened by atsakiridis about 9 years ago
#23 - HTML5/WebRTComm.js==>MMS on WSS: SSLException: Received fatal alert: protocol_version
Issue -
State: closed - Opened by yassen2gx about 9 years ago
- 4 comments
#22 - Remove the 'sip:' from the hello world js code
Issue -
State: closed - Opened by atsakiridis about 9 years ago
#21 - Update reference documentation
Issue -
State: closed - Opened by atsakiridis about 9 years ago
#20 - Introduce sample application
Issue -
State: closed - Opened by atsakiridis about 9 years ago
#19 - If media isn't setup properly in a call we should disconnect
Issue -
State: closed - Opened by atsakiridis over 9 years ago
- 2 comments
#18 - Failed dial to '1235' from Olympus from FireFox (Chrome works fine)
Issue -
State: closed - Opened by deruelle over 9 years ago
- 1 comment
Labels: bug
#17 - Proposed solution to #10
Pull Request -
State: closed - Opened by atsakiridis over 9 years ago
- 1 comment
#16 - Expose Twilio compatible 'Client' API
Issue -
State: closed - Opened by atsakiridis over 9 years ago
- 1 comment
#15 - Attended Call Transfer in WebRTCom
Issue -
State: open - Opened by yassen2gx over 9 years ago
- 4 comments
Labels: enhancement
#14 - ACK Issue with WSS Calls
Issue -
State: closed - Opened by deruelle over 9 years ago
Labels: bug
#11 - WebRTCom client ==>Asterisk==>WebRTCom client :ICE Iissue breaks call
Issue -
State: closed - Opened by yassen2gx over 9 years ago
- 6 comments
#10 - Make sure that m-lines count in the SDP answer are as many as in the offer
Issue -
State: closed - Opened by atsakiridis over 9 years ago
- 5 comments
#9 - IceServers parameter should use new "urls" format
Issue -
State: open - Opened by jaimecasero over 9 years ago
- 3 comments
#8 - Added data channel events for the app. Adjusted conditions to allow
Pull Request -
State: closed - Opened by jaimecasero over 9 years ago
Labels: enhancement
#7 - JainSip + WebRTComm to WEBRTC2SIP GW of Doubango : pb with Firefox
Issue -
State: closed - Opened by deruelle over 9 years ago
Labels: bug
#6 - CANCELing a call leads to timeout error on callee side
Issue -
State: open - Opened by deruelle over 9 years ago
- 1 comment
Labels: bug