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

#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

#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

#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

#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

#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

#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

#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

#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

#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

#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

#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