Ecosyste.ms: Timeline
Browse the timeline of events for every public repo on GitHub. Data updated hourly from GH Archive.
sipsorcery created a comment on a pull request on nofrixion/moneymoov-dotnet
Already approved but got merged to master instead of develop.
sipsorcery opened a pull request on nofrixion/moneymoov-dotnet
Improve payment request success URL parsing.
sipsorcery pushed 1 commit to master nofrixion/moneymoov-dotnet
- Revert "Improve payment request success URL parsing. (#453)" (#454) This reverts commit e966d6c3da6bce34d7cb8e120ab8... 8ae6cc9
sipsorcery closed a pull request on nofrixion/moneymoov-dotnet
Revert "Improve payment request success URL parsing."
Reverts nofrixion/moneymoov-dotnet#453
sipsorcery opened a pull request on nofrixion/moneymoov-dotnet
Revert "Improve payment request success URL parsing."
Reverts nofrixion/moneymoov-dotnet#453
sipsorcery created a branch on nofrixion/moneymoov-dotnet
revert-453-moov3928-payreq-succ-parsing - .NET Core SDK for NoFrixion's MoneyMoov API
sipsorcery pushed 1 commit to master nofrixion/moneymoov-dotnet
- Improve payment request success URL parsing. (#453) e966d6c
sipsorcery closed a pull request on nofrixion/moneymoov-dotnet
Improve payment request success URL parsing.
sipsorcery opened a pull request on nofrixion/moneymoov-dotnet
Improve payment request success URL parsing.
sipsorcery created a branch on nofrixion/moneymoov-dotnet
moov3928-payreq-succ-parsing - .NET Core SDK for NoFrixion's MoneyMoov API
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There's no way to tell at the SIP signalling level. The SIP traffic is identical no matter who or what picks up the call. The only option that occurs to me is some kind of analysis of the audio ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Determine if Call Was Answered by Person or Voicemail System
How can I detect if a call went to voicemail or was picked up by a person? I noticed that callResult still shows success, and IsCallActive remains true even when no one actually picks up. I conside...
sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery closed an issue on sipsorcery-org/sipsorcery
FfmpegToWebRTC Latency problem
I am using the FfmpegToWebRTC example and no matter what I try, I seem to always have about a 1.5 second delay. Currently using the following ffmpeg command: ffmpeg -re -rtsp_transport tcp -hwaccel...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Receiving Substreams from Simulcast-Enabled Devices Without Simulcast Implementation
We are currently utilizing Janus as our media server. Our setup involves joining a video room with an SDP generated by SipSorcery. When another device transmits video with simulcast enabled, we ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Getting `profile-level-id` in `OnVideoFormatsNegotiated`
I am having issues with Firefox peer, probably because it only supports baseline H.264 profile. I could tune down my encoder to provide that, but the issue is that on the SIPSorcery side `OnVide...
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- Changed the logging for the SIP and RTP UDP channels to omit the stack trace for expected socket closed exceptions. (... 1c6b956
sipsorcery created a branch on sipsorcery-org/sipsorcery
udp-channel-logs - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Added to enhancments tracking issue.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Data channels quickly slow down to a crawl under pressure
Please see https://github.com/sipsorcery-org/sipsorcery/pull/1087 for the test project. P.S. I think a unit test for transmitting a few GiB of data should be put in place.
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Slightly late reply. The SIPTranpsort class did have the facility to add transactions at one point, the commented out code can be seen [here](https://github.com/sipsorcery-org/sipsorcery/blob/e6...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
There seems to be no way of handling a received SIPNonInviteTransaction asynchronously
I receive a SIP (non-INVITE) request via `SIPTransport.SIPTransportRequestReceived` and would like to use `SIPNonInviteTransaction` but handle the transaction asynchronously. This leads into the pr...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The fix of hardcoding the SSRC value won't work. That's a pseudo random value and is expected to change for every RTP session. The cause of the issue here is why the RTP packet received on the m...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
OnDtmfTone triggered multiple times
HI, when I call my program from a mobile phone, OnDtmfTone is triggered 7 times. When i call my program from deskphone, OnDtmfTone is correctly triggered 1 times. Example from deskphone re...
sipsorcery reopened an issue on sipsorcery-org/sipsorcery
Crash in G722Codec.cs when called with odd number of shorts
Need some check in G722Codec.cs Encoder as crash at line 417 state.QmfSignalHistory[23] = inputBuffer[j++]; if inputbuffer has odd length.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
G729 encode problem
I use sipsorcery to answer a sip audio call with G729 codec. On sipsorcery side listen audio normally, but on caller side hear the noise(can hear volume change on speaking).