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sipsorcery

Rating
1481.30 (4,491,634th) 		Reputation
23,334 (5,592nd)
Page: 1 2 3 ... 11
Title Δ
How to add user-defined flags in Google WebRTC? 0.00
How should i pass a localStream object as an argument to an rtcpeer... 0.00
P2P Data transfer in Unity via WebRTC 0.00
INVITE request using SIP-Sorcery c# 0.00
Webrtc: What is the difference between onIceConnectionChange and on... 0.00
How to set framerate for IMFSourceReader for webcam capture on Wind... 0.00
Do I need SIP + WebRTC 0.00
Is it possible to successfully negotiate a WebRTC connection betwee... +0.53
SIP and WebRTC based browser client without SIP Registration 0.00
FFmpeg conversion RGB to YUV420 to RGB wrong result 0.00
Simple SIP call receiver and recorder 0.00
Setup SipSorcery Server 0.00
How to make a deadlock safe async library method -0.94
How to use a picture as a radio button in WPF? 0.00
Copy only changed and not null properties between two classes to a... 0.00
How to Run WireShark Commands in command line through C# -0.19
.NET/C#/Xaml display video from bytes or Bitmap 0.00
Capture camera stream via SDK that requires IntPtr to Window in WPF... 0.00
Is user part in sip 'Contact' header madatory? 0.00
Windows media foundation decoding audio streams -0.02
Understanding the webRTC gateway to connect SIP server 0.00
Where is the called party indicated? +0.53
VOIP/SIP Call Hold related questions 0.00
Windows Media foundation: IMFSinkWriter WriteSample API returning e... -0.75
stream audio from browser to WebRTC native C++ application 0.00
is SIP:480 after 180 ringing same as SIP:480 after 183? 0.00
Webcam source to EVR sink -0.51
Connect SIP with webRTC +0.26
How does a Symmetric CGN handle this scenario? 0.00
Weird network issue +0.53
How to send G722 encoded audio over an RTP stream? 0.00
What is the difference between the INVITE field and the To field in... -0.21
How to config Cisco 7941 IP Phone to work with VOIP (soft v.41.9-4) 0.00
PRACK Request using SIPSorcery 0.00
Webrtc-sip codec transcoding 0.00
How to hold SIPSorcery VOIP call in C#? 0.00
how to find from RTP stream that it belong to which sip call? +0.03
sending SIP messages with C# 0.00
re-transmission of re-invite over UDP 0.00
Asterisk / SIP - Delaying the 200 OK till after calling card is ent... +0.04
SIP dialog CSeq sequence number 0.00
In a SIP INVITE Method Getting 200 Response continuosly +0.01
How to validate WebRTC connection signals when peers can't trus... 0.00
WebRTC, fake ICE server for one-to-one (client-server) connection +0.01
WebRTC P2P stream a youtube video +0.53
How to implement HTTP protocol using WebRTC? +2.25
WebRTC and security - intercepting connections? 0.00
Use Media Foundation H.264 encoder with live555 0.00
How to send webcam stream to wowzaurl using webrtc 0.00
What is the status of Microsoft Media Foundation? +0.53