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
- Removed redundant ffmpeg init call. e64ae70
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Yes, needs to be set when you create teh FFmpegVideoEndPoint, or equivalent.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
RTCRtpEncodingParameters.maxBitrate
I might have missed this issue #346. But simply put, is there a way to set `RTCRtpEncodingParameters.maxBitrate` in `sipsorcery`?
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Fixed in #1225. Also make sure to run: `winget install "FFmpeg (Shared)" --version 7.0`
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Cannot get WebRTCReceiver example to run
Hi, I'm trying to do bidirectional video streaming between a C# application running on Windows 10 and Chrome (v98) on Android 10. I have cloned the latest repo, changed the "version" tag in glob...
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- Updated webrtcreceiver dotnet version and nuget packages and added mssing ffmpeg init command. (#1225) 7071898
sipsorcery created a branch on sipsorcery-org/sipsorcery
fix-webrtcreceiver - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery closed an issue on sipsorcery-org/sipsorcery
After switching device, loosing audio/video
If I switch from one device to another (mic to speaker OR one video source to another video source) loosing audio/video track. How to resolve this issue?
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
SRTP has only been implemented for WebRTC. It has not been wired up for SIP.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SRTP SDP processing for SIP
Hello, we've been working on a project to emulate SIP Devices using SIP Sorcery. Our project supports only SRTP SDP and not RTP. We tried to add media in our project using VoIPMediaSession class. ...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There has been a number of improvements in RTCP processing since this issue was originally created. If there is still interest and the problem still exists please feel free to re-opne.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Understanding the Data Channel Logs
So, attached are two log files ([left.log](https://github.com/sipsorcery-org/sipsorcery/files/9234874/left.log) [right.log](https://github.com/sipsorcery-org/sipsorcery/files/9234875/right.log)) fo...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
I realise this response is very late. If you're still interested and able to provide the library debug messages please feel free to re-open.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
UserAgent.Hangup() Sometimes couldn't Hangup an established call.
I use SIPUserAgent to make a call。But Sometimes it may HangUp fail. I can't find why and when it occurred. the code below is my makecall code: public async Task MakeCall(string desti...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SIPSorcery.SIP.App.SIPUserAgent.ClientCallAnsweredHandler causes an ArgumentNullException
I found an unexpected exception thrown by SIPSorcery.SIP.App.SIPUserAgent.ClientCallAnsweredHandler, caused application crash, when my server application starts a phone call sometimes. Below is th...
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- Add a handler for SIP UAC getting an INVITE response with no SDP. (#1224) d36d795
sipsorcery closed a pull request on sipsorcery-org/sipsorcery
Add a handler for SIP UAC getting an INVITE response with no SDP.
sipsorcery opened a pull request on sipsorcery-org/sipsorcery
Add a handler for SIP UAC getting an INVITE response with no SDP.
sipsorcery created a branch on sipsorcery-org/sipsorcery
uac-handle-noremotesdp - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SIPTransport.SIPTransportRequestReceived async event result not checked
Hi, as you may see at https://github.com/sipsorcery-org/sipsorcery/blob/afb3cd3b7f0f05df109aaea4faa9b23650902803/src/core/SIP/SIPTransport.cs#L1080 a task returned from `SIPTransportRequestRe...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Buffering on the WebRTC
Is there any way to switch off totally buffering on the WebRTC side? Im using the H.264 under the hood with "zerolatency" option but anyways I think that jitter is comming to play on the WebRTC sid...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Not sure what the issue was at the time but SIP DNS lookups are working fine now: ``` c:\dev\sipsorcery\examples\sipcmdline>dotnet run -- -d [email protected] -v -s uac [22:50:59 DBG] RunCommand...