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
Issue with Audio Level RTP Extension in Janus-Gateway
I'm currently working with Janus-Gateway and I’ve encountered an issue where I’m not receiving any talking events. After some troubleshooting, I was asked whether the audio level RTP extension is b...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Cannot detect when a track is added by the remote peer
Similar to #983. I am attempting to add a media track some time after the peer connection has been established (this code is executed by the local peer): ``` var localVideoTrack = new MediaS...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Add support for RFC3311-UPDATE method
I have a voice call, and the call will be automatically disconnected 10 minutes after it is answered. By checking the signaling, I found that the third-party PBX（freeswitch、VOS3000 etc） will sen...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Punch multiple holes and exploit the Birthday Problem
Is there any way to create `n` random candidates, even if it's the same IP and peer type, for each peer type (with presumable different port numbers)? NAT hole-punching seems to have worked much be...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Datachannel Reliability Implementation
Currently `RTCDataChannelInit` doesn't appear to have any effect: https://github.com/sipsorcery-org/sipsorcery/blob/c153907c3d478e179974ed8aa426233be5a7aeaa/src/net/WebRTC/RTCDataChannel.cs#L92 htt...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Select an exact VideoFormat/AudioFormat during Webrtc SDP
I would like to use an exact VideoFormat (i.e `Name H264, FormatID 125, Codec H264, ClockRate 90000, Parameters level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f` ) during SDP ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
RFC 7714 (AES-GCM AEAD support)
Hi all, I'm currently working on a project and I need to secure the connection using a cryptography with 256-bit key. Reading the specifications, I found that original SRTP only require 128-bit al...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Bandwidth estimation
And one more thing from my side as I can't see that happening in action, but may be wrong. Does Sipsorcery implement bandwidth estimation and while having active session adjust the bitrate of encod...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SIPUserAgent.Call can now be called with the addition AuthName optional parameter.
Added Optional Parameter to function Filename: SIPUserAgent.cs public Task<bool> Call(string dst, string username, string password, IMediaSession mediaSession, int ringTimeout = 0, string optio...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
STUNClient.cs STUNAttributeTypesEnum.XORMappedAddress not handled
I noticed that both stun.3cx and stun.google returned XORMappedAddress but public static IPEndPoint GetPublicIPEndPoint(string stunServer, int port = DEFAULT_STUN_PORT) doesn't seem t...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SRTP only works with a single media type at a time
When using the newly added SRTP implementation for SIP calls, it seems that the media only works for one of the media types at a time. For example, if I create a media call with audio AND video, ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
VP9 Library
I'm not sure if this is useful or not but I think this might be a full VP9 decoder written in c# that you might find interesting. https://github.com/Ryujinx/Ryujinx/tree/master/Ryujinx.Graphics....
sipsorcery closed an issue on sipsorcery-org/sipsorcery
.NET Framework timing problem
In order to avoid .NET Framework timing problem please evaluate if you find useful to replace (or to create a new object) all timers of AudioExtraSource with [MicroTimer](https://www.codeproject.co...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Add proper RTCP feedback support as per RFC4585
[RTCP Feedback](https://tools.ietf.org/html/rfc4585) provides a way to deal with dropped or missing RTP packets. For voice a dropped RTP packet is not worth taking any action over. Each voice ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Optimisations required for 1080p
Attempting to receive a VP8 encoded video stream at 1080p (1920x1080) will quickly degrade to become very laggy and then corrupted. The sipsorcery library is not able to decode the video stream fas...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Add support for the Google Congestion Control Algorithm
Currently when Chrome is requested to send a 1080p stream to an application using the sipsorcery library the frame resolution is ramped up over a period of up to 30s, e.g: ```` [21:02:57 DBG] A...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Add support for WebRTC session re-negotation
As per the [Session Negotiation Model](https://w3c.github.io/webrtc-pc/#session-negotiation-model). Also see https://blog.mozilla.org/webrtc/perfect-negotiation-in-webrtc/ for additional context.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Add H264 depacketisation
Currently the `RTPSession` class supports `H264` packetisation [but not depacketisation](https://github.com/sipsorcery/sipsorcery/blob/f1d0b3d0994bd1feb967c278ced2b3ad1dc05881/src/net/RTP/RTPSessio...
sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
All packages have been updated in the latest releases.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
add_OnVideoSourceRawSampleFaster not implemented when cloning and running WebRTCGetStarted
When cloning the repo and running the [WebRTCGetStarted](https://github.com/sipsorcery-org/sipsorcery/tree/master/examples/WebRTCExamples/WebRTCGetStarted) example, if you press 'start' in the web ...
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