Ecosyste.ms: Timeline
Browse the timeline of events for every public repo on GitHub. Data updated hourly from GH Archive.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Remote ICE candidate only supports multiplexed media
Trying to send a video stream from Unity WebRTC to SIPSorcery.Net using the WebSocketRTCPeer from the examples. When exchanging candidates, onIceCandidateError throws the following exception: ...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There have been some DTLS improvements recently that may help with this issue, If the problem is still occurring with the latest code please feel free to re-open this issue.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
WebRTC DataChannel shuts down(close) after some time of operation
Greetings, everyone. I am using DataChannel to transmit frames. After an indefinite amount of time (always different times, but within 10 minutes), my DataChannel with whoever is broadcasting is in...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
I did implement some changes to the DTLS cipher suite selection as well as switched the default signature algorithm to ECDSA. While it does not directly address the cause of the "bad mac" alert it ...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The RTP timestamp does [get intialised](https://github.com/sipsorcery-org/sipsorcery/blob/de91be67f8c79d6fd67bb5b18a7f9f33bd9ba6f5/src/net/RTP/RTPHeader.cs#L60) to a pseudo-random value.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Initial RTP packet's timestamp should be random (RFC 3550)
As per RFC 3550, the initial timestamp on an RTP stream should be random. However, according to .\net\RTP\MediaStreamTrack.cs, the timestamp of the initial RTP packet always defaults to zero. >...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
I suspect this could be an interop issue between dotnet and SDL2 which is not directly related to this library.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
the application works in debug mode but crashes in non-debugging mode
Application: ssss.exe CoreCLR Version: 7.0.923.32018 .NET Version: 7.0.9 Description: The process was terminated due to an unhandled exception. Exception Info: System.AccessViolationException: ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
VP8: Crash on Video Input from an Android Device
Using last version of SIPSorcery (master branch) and SIPSorcery.FFmpeg v1.0.0 I have always a crash **when decoding a input video frame from an Android device.** **Sending a video stream in VP8...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There have been some improvements in the WebRTC negotiation and DTLS handling that may have fixed these problems. If not please feel free to re-open this issue.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
WebRTC peer connection keeps closing
I've done my implementation based on [FfmpegToWebRTC example](https://github.com/sipsorcery-org/sipsorcery/tree/master/examples/WebRTCExamples/FfmpegToWebRTC) using .NET 6 and latest SIPSorcery (6....
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
> Is there an option to turn IPv6 off as a workaround? Yes, the config below will force the underlying RTP socket to bind to the IPv4 wildcard address and not use IPv6. ``` RTCConfiguration ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
private address cannot be parsed correctly
 I met a problem on one of my computer with win 7, the private address cannot be pa...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Hopefully this has been fixed in .net8 and in the interim some object disposed exception handlers have been added.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Socket.EndReceiveFrom will never be called
EndReceiveFrom() will never be called here, because AsyncCallback is null: https://github.com/sipsorcery-org/sipsorcery/blob/96ea5d7e81406a077c7571819f07f36ab39b7e89/src/sys/Net/NetServices.cs#L50...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SIPSorceryMedia.Encoder BadImageFormatException/EntryPointNotFoundException
Hi Experts, I am using SIPSorceryMedia.Encoder for streaming image frames as video and facing following problem. 1. Nuget manager is adding vpxmd.dll x64 version for SIPSorceryMedia.Encoder wh...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Test fails (Debug/AnyCPU)
SIPSorcery.Net.UnitTests.SctpDataSenderUnitTest.SmallBufferSend Source: SctpDataSenderUnitTest.cs line 39 Duration: 1 sec Message: The collection was expected to contain a single el...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Seems the [Prack popery](https://github.com/sipsorcery-org/sipsorcery/blob/de91be67f8c79d6fd67bb5b18a7f9f33bd9ba6f5/src/app/SIPUserAgents/SIPUserAgent.cs#L391) for the SIPUserAgent is in place now.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SipUserAgent disable PRACK support
Hi, I would like to disable disable PRACK support when using the SIPUserAgent. I've identified the following changed needs to be done to the library, correct me if I'm wrong: - SipUserAg...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Suspect this is/was something to do with the need to support SDP re-negotiation which has not yet been implemented.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
sender.replaceTrack from chrome cause connection to drop.
@sipsorcery looks like chrome send RTCP BYE event when replacing track using sender.replaceTrack (switching camera to screen share) In SipSorcery when RTCP BYE was received the connection w...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There have been some improvements in the WebRTC peer negotiation logic over the last few years. If you're still interested and still have this issue with the latest code please feel free to re-open...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
ICE Infinite loop for WebRTC Connection
Hello, when testing webRTC, the connection was closed from one peer but another was always connecting and seemed stuck. When debugging I found it was checking ice state in infinite-loop (snaps...
sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Sorry for the delayed response. DTMF does not get a lot of love these days. More than happy to take a PR if you're still interested. Your approach sounds sensible.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Transmitting in-band RTP events and audio packets on a media stream
Hi, Thanks for all your work on sipsorcery. **Summary** I noticed that SendDtmfEvent was moved out of RTPSession.cs and into AudioStream.cs since v6.0.8. With this change, I think it has b...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There have been a number of improvements to the WebRTC peer negotiation that may have fixed this issue. If you're still interested and it's still a problem please feel free to re-open this issue,
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Race condition - Webrtc Ice connection fails sometimes
I use Sipsorcery for connecting to a remote Webrtc peer (mobile Chrome browser) at the same WIFI network. The connection with Sipsorcery sometime fails (Race condition) if I enable VPN client at...
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
If you use the [Call method](https://github.com/sipsorcery-org/sipsorcery/blob/de91be67f8c79d6fd67bb5b18a7f9f33bd9ba6f5/src/app/SIPUserAgents/SIPUserAgent.cs#L485) that takes a SIPCallDescriptor yo...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Cannot pass message content while initiating sip call
As per RFC we are allowed to send different headers and body with SIP invite. In this case we can pass content type as multipart/mixed and body consisting of multiple contents separated by boundary...