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...
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...
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...
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 ...
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...
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:
...
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.
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...
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 ...
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.
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.
>...
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: ...
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...
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.
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....
> 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 ...
![image](https://user-images.githubusercontent.com/26663548/162912644-12df18db-1022-4b3a-b757-8e4cc940d98b.png)
I met a problem on one of my computer with win 7, the private address cannot be pa...
EndReceiveFrom() will never be called here, because AsyncCallback is null:
https://github.com/sipsorcery-org/sipsorcery/blob/96ea5d7e81406a077c7571819f07f36ab39b7e89/src/sys/Net/NetServices.cs#L50...
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.Net.UnitTests.SctpDataSenderUnitTest.SmallBufferSend
Source: SctpDataSenderUnitTest.cs line 39
Duration: 1 sec
Message:
The collection was expected to contain a single el...
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.
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
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...
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...
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...