Ecosyste.ms: Timeline

Browse the timeline of events for every public repo on GitHub. Data updated hourly from GH Archive.

sipsorcery-org/sipsorcery

sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery

View on GitHub

sipsorcery created a tag on sipsorcery-org/sipsorcery

v8.0.3 - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.

sipsorcery closed an issue on sipsorcery-org/sipsorcery
ice-mismatch when calling an UA based on PJSIP
Hi, I'm trying to call a SIP client based on PJSIP, but the SDP negotiation fails because the PJSIP client answer with an attribute a=ice-mismatch. I think I should modify the offer sent, but...
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
  • Disabled another failing test on macosx. eae2fa7

View on GitHub

sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
  • Fixed logging in webrtc get started example. 08b9cb0
  • Update README.md f349f0f
  • Merge branch 'master' into gh-pages b6ab521

View on GitHub

sipsorcery pushed 2 commits to master sipsorcery-org/sipsorcery
  • Bumped version to 8.0.2. cc829db
  • Disabled some failing unit tests for macosx. 4b934f6

View on GitHub

sipsorcery created a tag on sipsorcery-org/sipsorcery

v8.0.2 - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.

sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery

View on GitHub

BlueSharkPartners starred sipsorcery-org/sipsorcery
paigewoodmann starred sipsorcery-org/sipsorcery
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
  • Fixed logging in webrtc get started example. 08b9cb0

View on GitHub

sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Ok, thanks for testing. Seems it's something on macOSX and not much that can be done here.

View on GitHub

sipsorcery closed an issue on sipsorcery-org/sipsorcery
Connections dropping on OSX Sequoia
Using the latest version of OSX (sequoia 15.1 beta and release channel) - connections work fine on same computer but connecting over the network or internet ICE completes, connection succeeds but t...
ispysoftware created a comment on an issue on sipsorcery-org/sipsorcery
Yeah that didn't help unfortunately - if that was the problem it wouldn't be working fine via the terminal. It's definitely a bug in macOS but they seem to be refusing to acknowledge it https://d...

View on GitHub

sipsorcery closed an issue on sipsorcery-org/sipsorcery
Make a call using SIPWebSocketChannel towards a registered WebRTC client
How can we make an outbound call using the SIPWebSocketChannel towards a registered WebRTC client?
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The [SIPCallServer](https://github.com/sipsorcery-org/sipsorcery/blob/master/examples/SIPExamples/SIPCallServer/Program.cs) is the closest example.

View on GitHub

sipsorcery closed an issue on sipsorcery-org/sipsorcery
How to realize the ip phone proxy server function similar to opensips
Hello, I want to realize the intranet IP phone inter-dial function, similar to opensips, I saw that the REGISTER function is implemented in the projects UserAgentServer and SIPProxy in SIPExamples,...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Not receiving RTP events or reports
I am streaming from my server to the browser using webrtc and FFMPEG. I am trying to figure out if I am getting packet loss indicators from the browser and then trigger a keyframe from my encoding ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
"Option not found" with encoder option -vf scale=1280:720
hello , thanks for this great library , it helps me alot , but i am facing an issue , i am trying to scale down the frame that i am passing to ffmpeg using encoderOptions in FFmpegVideoEncoder thr...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
When Re-joining to janus audio bridge, peer connection could not stablished
STEP 01: Successfully joined in janus audio brigdge room and audio call is on. STEP 02: Left from Room STEP 03: after joining again, getting following status Ice State has Changed check...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
That's by design. The value returned by the server is the value it wants the client to use. If you use a larger value the server will discard the registration once its expiry period is reached.

View on GitHub

sipsorcery closed an issue on sipsorcery-org/sipsorcery
Defect in REGISTER Expire value
Hi, I think I've found a defect in the value of the `Expires` header in the `REGISTER` message. If the registrar server adds the `expires` parameter in the `Contact` field in the `OK` respons...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Unable to create WindowAudioEndPoint
I have observed the exception when unable to create the WindowAudioEndPoint when working with NAudio. Code: ` var userAgent = new SIPUserAgent(sipTransport, OUTBOUND_PROXY); userAgent.ClientC...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
audio no send any 27 calls
Hello Master, I need help to understand why I have this exception, it ends my application, what I see is that it generates when the audio is sent! Error 9/13/2021 12:49:12 PM Application Er...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
AudioExtrasSource - object disposed exception in SendStreamSample
Running version 5.2.3 in a load test scenario. We had an issue where the SIP switch (Freeswitch) was sending a BYE just after accepting a call, causing the media stream being played to be closed r...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The OS is responsible for controlling access to the aduio devices.

View on GitHub

sipsorcery closed an issue on sipsorcery-org/sipsorcery
Microphone and speaker sound problems.
When I simultaneously transmit the following media: Output screen. Output speaker sound. Output microphone sound. When using webrtc.html to play on the browser side, the following problems oc...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There is defintiely response retry logic present already. If you capture the SIP exhange using WireShark you'll see 11 retries.

View on GitHub

sipsorcery closed an issue on sipsorcery-org/sipsorcery
SipServerUserAgent.Answer is missing retransmit logic for invite responses?
Hi, I'm using the SipUserAgent class to respond to invites from the server. The happy case of course works as expected. It responds with trying, ringing and once answered it responds with an OK ...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
That's a weird error. WebRTC only supports multiplexed media and the media id 0, media id 1 etc are used to identify sections in the SDP to describe different media types. AFAIK there's no way to c...

View on GitHub

Load more