Ecosyste.ms: Timeline
Browse the timeline of events for every public repo on GitHub. Data updated hourly from GH Archive.
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- Added new nuget gha and updated examples gha. e632063
sipsorcery pushed 2 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery created a tag on sipsorcery-org/sipsorcery
v8.0.4 - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
ispysoftware created a comment on an issue on sipsorcery-org/sipsorcery
For anyone unfortunate enough to end up here it needs to be run as a launch agent not as a launch daemon and the application needs to be in the Applications folder - it can't be on the desktop.
ispysoftware created a comment on an issue on sipsorcery-org/sipsorcery
@sipsorcery do you think it could be anything similar to this? https://github.com/netty/netty/issues/11563 https://github.com/netty/netty/pull/12019/commits/6d90b807d55794f5c1970344d743b618...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The trick here is to use the same `SIPTransport` instance when you create the two different user agents. That way they will both use the same UDP port to send their requests from and will end up wi...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SipSorcery Register and Invite use different ports
I am currently trying make a call with a small WPF application on my windows pc via [SipSorcery](https://github.com/sipsorcery-org/sipsorcery) . I am using .net 6 and latest version of SipSorcery p...sipsorcery closed an issue on sipsorcery-org/sipsorcery
Set protocol (tcp) for RTC Ice Candidate
If I create a new instance of [RTCIceCandidate](https://github.com/sipsorcery-org/sipsorcery/blob/c8fee0eebc27a2e73b612728e9851b8f420e2223/src/net/ICE/RTCIceCandidate.cs#L118) from `RTCIceCandidate...sipsorcery closed an issue on sipsorcery-org/sipsorcery
Switching video source to a different resolution/aspect ratio
We are trying to switch video source `(IVideoSource`) on a live `MediaSteamTrack `(`SendOnly`) in an open and connected `RTCPeerConnection` and the two video sources have different aspect ratios. ...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The console logs, or even better SIP trace, are needed to look into this issue.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
First, call out failed, and then the incoming call was answered. The SIPUserAgent.OnCallHungup Event was not triggered when hanging up
1. Call out failed, the Event "private void ClientCallFailedHandler(ISIPClientUserAgent uac, string errorMessage, SIPResponse sipResponse)" was not invoke the "CallEnded" event, so the CallDescript...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There is no proper RTCP feedback mechanism in the sipsorcery library and minimal work has been done to wire up the video encoders to trigger key frames in the event of packet loss. I'm adding this ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
H264 bad video after passing through network; issue with packetization/depacketization apparently..
I am trying to fix an issue that comes from sending the Android camera using SipSorcery, after several tests I ended up recording the stream in raw h264 just before sending it and just when receivi...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
As per #728 Chrome uses an SSRC of 1 in RTCP Receiver Reports when it isn't sending a stream. The SSRC in the Receiver Report does match the SSRC of the stream it's receiving from sipsorcery.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Why Chrome is not considering the SSRC that sipsorcery sends?
So, running the `WebRTCGetStarted` example using the latest version of the library. 1. The SDP the library sends to Chrome contains the following lines: `a=ssrc:978236180 cname:bbbc2e1c-fdc7-436...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Checked in WireShark and Chrome does use SSRC of 1 when it's not sending anything. This is from teh WebRTCGetStarted example where the audio & video is sendonly to the browser. ![image](https://...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
RTCP Compound Packets broken for WebRTC connections
I am finding that the RTCP Compound packets for WebRTC connections are causing issues in the current master. By the looks of it the RTCP Compound packets data is trying to be parsed and the dat...sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- Un-obsolete STUN address attribute. Fix warnings. (#1229) 6ae72de
sipsorcery closed a pull request on sipsorcery-org/sipsorcery
Un-obsolete STUN address attribute. Fix warnings.
sipsorcery opened a pull request on sipsorcery-org/sipsorcery
Un-obsolete STUN address attribute. Fix warnings.
sipsorcery created a branch on sipsorcery-org/sipsorcery
fix-warnings - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.